hash stringlengths 32 32 | doc_id stringlengths 5 12 | section stringlengths 5 1.47k | content stringlengths 0 6.67M |
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1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 7.2.2.2 Error correction facsimile data | As these facsimile coded data between the fax adaptor and the facsimile terminal are structured in HDLC frames, the handling of this procedure segment will exploit such formatting. The content of such an HDLC frame is further on called a block.
Each such block is included in the information field of a error correction data element of the FA protocol which is processed for transmission across the radio interface as outlined in clause 6.
The message phase (see figure II.8/GSM 03.46) at both the PLMN ends is triggered by the transit of a confirmation frame (CFR, MCF, PPR, CTR or ERR) sent by the receiving terminal and marking the end of the BCS phase.
If four consecutive PPR are counted within the same "partial page", the BCS phase continues.
The transmitter adaptation function will enter the message phase as per CCITT Recommendation T.30 standard procedure.
The terminal adaptation function associated with the receiving terminal after receiving facsimile coded data or autonomously 5.5s after detecting the trigger frame will change the modem function to V.27ter or V.29 CCITT Recommendation and initiate the training at the applicable speed.
Following the training segment, HDLC flags will be stuffed towards the facsimile terminal until a FCD frame is detected, that will mark the beginning of the real phase C.
If due to a preceding error the message phase cannot be entered, this training must be aborted when a new BCS element is received by the transmitting fax adaptor. |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 7.2.2.3 Buffering of facsimile coded data | The following subclauses only apply, when using the normal facsimile data transfer, i.e. not with the error correction mode. |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 7.2.2.3.1 Transmitter adaptation function | In the transmitter adaptation function the facsimile coded data being received from the facsimile terminal are transcoded stripping of FILL information and written into the buffer.
If there is enough information available, this data is read out from the buffer, and a FA protocol element is generated which is processed as described in clause 6 to be transferred to the receiver adaptation function using one of the standard TAFs referred to in subclause 7.1. For that purpose the data is segmented in blocks (see subclause 6.2.5.2).
Due to the ARQ techniques of the RLP the throughput across the radio interface may be less than the message speed between the transmitting facsimile terminal and the transmitter adaptation function, i.e. the content of the buffer may increase. When a certain threshold is reached from which the fax adaptor can derive that the actual page cannot be transmitted successfully, the connection may be prematurely released.
If the throughput at the radio interface is greater than the message speed between the transmitting facsimile terminal and the transmitter adaptation function (e.g. when the end‑to‑end speed is lower than 9 600 bit/s), the buffer may be empty most of the time. |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 7.2.2.3.2 Receiver adaptation function | In the receiver adaptation function FILL information is transmitted to the facsimile terminal at the beginning of each page, if necessary, to bridge the gap between the training sequence and the real facsimile coded data. In case of normal fax data the FILL 0's can be expanded up to 5s only and therefore after these up to two white scan lines should be inserted, if necessary.
The facsimile coded data received across the radio interface are re‑generated from the LAPB, L2R and FA protocol elements, reversely transcoded according to the knowledge of the fax adaptor, and written into the buffer. The reverse transcoding consists of insertion of FILL information before the facsimile coded data is forwarded to the facsimile terminal to comply with the recognized minimum line length as defined in CCITT Recommendation T.4.
At the beginning of each page the facsimile coded data to be sent to the facsimile terminal is not read out from the buffer until at least 2 instances of EOL or an RTC have been received or the following buffer size limit, depending on the end to end data transfer rate, has been exceeded:
2 kByte for 2,400 bit/s;
4 kByte for 4,800 bit/s;
6 kByte for 7,200 bit/s;
8 kByte for 9,600 bit/s.
Once this procedure has been started, i.e. during the page transmission, the facsimile coded data is transmitted, however, the following EOL is delayed by inserting additional FILL information, if necessary, until the pre‑set threshold (2 EOLs or the buffer size limit) is reached again. If the actual coding line is going to exceed 5 s, the threshold is temporarily reduced, i.e. the following EOL is sent. However, the buffering algorithm shall try to reach the pre‑set threshold again as fast as possible (by inserting FILL also before following EOLs). If no EOL is available to be transmitted to the facsimile terminal for a period greater than 5 s, the connection will be released by an ordinary receiving facsimile terminal (ref. CCITT Recommendation T.4). |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 7.2.3 Disconnect procedure | The transmitter adaptation function, upon detection of the DCN frame (see CCITT Recommendation T.30) sent by the local terminal to indicate the end of the facsimile transmission, initiates the disconnect procedure. |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 7.2.4 Timeouts | The overall fax adaptation function is in principle bound to the timing constraints associated with the end‑to‑end CCITT Recommendation T.30 procedure. This means that, no matter of the reference configuration used at the mobile station, either the "standard" one (figure 2a/GSM 03.46) or the "GSM facsimile machine" (figure 2d/GSM 03.46), the progress of the call will be mainly subject to the CCITT Recommendation T.30 typical timing protections, settled externally.
However, due to the specific conditions caused by the GSM PLMN system, there is the need for a special support with respect to BCS command repetitions as explained above. For that purpose, the fax adaptors will provide means for local time‑out. The timer will be started and stopped as described in the applicable clauses of the CCITT Recommendation T.30. |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 8 Signalling aspects | GSM 07.03 identifies the bearer capability requirements to be supported by the terminal adaptation function in the MT (see GSM 07.01 for BC and HLC coding). The specific signalling requirements are those for "speech" and "facsimile group 3" or "facsimile group 3" only, respectively. The MT indicates in the call set up request the requirements, e.g. first speech, second facsimile by sending the bearer capability information element(s) in the appropriate order. For an "auto calling" facsimile request, the facsimile group 3 bearer capability is sent as the first or the only bearer capability for Teleservice 61 or 62, respectively.
For interworking between Teleservice 61 and Teleservice 62 refer to GSM 02.03 and 07.01. |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 8.1 Handling of tonal signals | Because the CCITT defined service uses modems, there are some signals received from the analogue link at the MSC/IWF and (where used) the fax adaptor which do not have a direct binary representation. These signals cannot therefore be passed across the radio interface in the same way as the CCITT Recommendation T.30 and CCITT Recommendation T.4 information.
These signals are the modem called (CED) and calling (CNG) tones sent at the start of each fax data phase of the call; they are generated locally by the FA/MT and/or FA/IWF, exploiting an end‑to‑end time alignment mechanism, triggered by appropriate messages on the GSM signalling channel. The procedure is detailed in the following. |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 8.2 Call establishment | |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 8.2.1 Mobile terminated call | The PSTN facsimile group 3 terminal may be manually or automatically calling. |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 8.2.1.1 Speech then facsimile | Refer to the diagram in figure II.1a/03.46 and II.1b/03.46. In both of the figures the initial call setup is mobile terminated. In figure I.1a/03.46 the DCD is also mobile terminated (MT), while the DCD in figure I.1b/03.46 is mobile originated (MO).
In order to make the transition from the speech phase to the facsimile phase, the MODIFY command must be initiated by MMI at the MS.
In the case where a GSM facsimile machine is used, it will turn on circuit 108/2 when it is connected to the line by manual intervention.
In the case where a fax adaptor at MT is used, it will turn on circuit 108/2, when the mobile fax apparatus is connected to the line by manual intervention.
After determination of the DCD and ICM (see subclause 6.4) and on completion of the synchronization process over the radio interface or the RLP establishment, CT107 shall be turned on by the MT; in case where a FA is used, on receipt of CT107 from MT, the FA will complete the tonal handshaking according to the rules in subclause 6.4.
The analogue link at the FA/IWF side will be established in accordance with the T.30 rec.; provided the synchronization process is completed (CT108.2 ON condition), the appropriate tone according to the rules in subclause 6.4 shall be transmitted. In case of DCD mobile terminated the CED tone shall be transmitted after a silence of 1.8 to 2.5 sec (see T.30, 4.3.3.2) from the call being answered; during transmission of CED tone (2.6 sec minimum duration, followed by a delay period of 75 +/‑ 20 ms) the FA/IWF will process data received from the GSM‑TCH as usual, but relevant information (e.g. preamble of a BCS frame) shall be discarded without any buffering. Note that circuit 109 and circuit 106 (according CCITT Recommendation V.24) at the R interface of the MT must be turned on by the fax adaptor at the IWF before any further procedure can be carried out between the fax adaptors and consequently end‑to‑end. Once the connection is established, both circuit 106 and circuit 109 are clamped to the ON condition by the fax adaptor at the IWF, so fixing a full duplex mode throughout the whole facsimile phase of the call. |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 8.2.1.2 Auto answer | Refer to diagram in figure II.2/GSM 03.46. A call received from the PSTN will cause the MT to turn on circuit 125 (according to CCITT Recommendation V.24) at the R interface.
In the case where a GSM facsimile machine is used, CCITT Recommendation V.25bis auto answering process is handled directly by turning on circuit 108/2.
In the case where a fax adaptor is used, circuit 125 will cause ring current to be sent to the mobile facsimile terminal. The fax adaptor will turn on circuit 108/2, when the mobile facsimile terminal answers the call.
On receipt of circuit 108/2, the MT will answer the call and initiate the synchronization process and the establishment of the RLP across the radio interface. On completion of the synchronization process or RLP establishment, the modem at IWF will automatically be selected and send CED to PSTN facsimile terminal. Also circuit 107 shall be turned on by the MT.
In the case where a fax adaptor is used, on receipt of circuit 107 from MT, the fax adaptor will initiate the tonal hand‑shake by sending CNG (mandatory).
The analogue links at both the PSTN side and the mobile side (where a fax adaptor is used) will be established in accordance with the appropriate V. series recommendation.
Note that circuit 109 and circuit 106 at the R interface of the MT must be turned on by the fax adaptor at the IWF before any further procedure can be carried out between the fax adaptors and consequently end‑to‑end. Once the connection is established, both circuit 106 and circuit 109 are clamped to the ON condition by the fax adaptor at the IWF, so fixing a full duplex mode throughout the whole facsimile phase of the call. |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 8.2.2 Mobile originated calls | The PSTN facsimile group 3 terminal may be manually or automatically answered. |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 8.2.2.1 Speech then facsimile | Refer to the diagram in figure II.3a/03.46 and figure II.3b/03.46. In both of the figures the initial call setup is mobile terminated. In figure II.3a/03.46 the DCD is also MO, while in figure II.3b/03.46 the DCD is MT.
In order to make the transition from the speech phase to the facsimile phase, the MODIFY command must be initiated by MMI at the MS, which will result in an establishment of the RLP across the radio interface and connection to line of the FA/IWF.
In the case where a fax adaptor is used, the mobile facsimile terminal must be connected to line by manual intervention at this stage, and will cause the fax adaptor to turn on circuit 108/2 (according to CCITT Recommendation V.24) towards the MT.
In the case where a GSM facsimile machine is used, circuit 108/2 shall be turned on when the GSM facsimile machine is connected to line by manual intervention.
After determination of the DCD and ICM (see subclause 6.4) and on completion of the synchronization process across the radio interface or the establishment of RLP, the modem at the IWF will be automatically selected and send the appropriate modem tone according to the rules in subclause 6.4 to PSTN facsimile terminal. Also circuit 107 shall be turned on by the MT, whereupon the FA/MT will complete the tonal handshaking according to the rules in subclause 6.4.
In the case where a fax adaptor is used, the receipt of circuit 107 shall cause the fax adaptor to connect to line.
The analogue links at both the PSTN side and the mobile side (where a fax adaptor is used) will be established in accordance with the appropriate CCITT V. series recommendation.
Note that circuit 109 and circuit 106 at the R interface of the MT must be turned on by the fax adaptor at the IWF before any further procedure can be carried out between the fax adaptors and consequently end‑to‑end. Once the connection is established, both circuit 106 and circuit 109 are clamped to the ON condition by the fax adaptor at the IWF, so fixing a full duplex mode throughout the whole facsimile phase of the call. |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 8.2.2.2 Auto calling | Refer to diagram in figure II.4/GSM 03.46. The auto calling procedure of CCITT Recommendation V.25bis is initiated at the CCITT Recommendation V.24 interface. This is done either directly from the GSM facsimile machine or, in the case where a fax adaptor is used, by loop disconnect or DTMF dialling information between the mobile facsimile terminal and the fax adaptor.
When the call is answered, the synchronization process will be started and the RLP will be established across the radio interface.
On completion of the synchronization process across the radio interface or RLP establishment, the modem at the IWF will be automatically selected and send CNG (mandatory) to PSTN facsimile terminal. Also CT107 shall be turned on by the MT.
In the case where a fax adaptor is used, the receipt of circuit 107 shall cause the fax adaptor to connect to line.
The analogue links at both the PSTN side and the mobile side (where a fax adaptor is used) will be established in accordance with the appropriate V. series recommendation.
Note that circuit 109 and circuit 106 (according to CCITT Recommendation V.24) at the R interface of the MT must be turned on by the fax adaptor at the IWF before any further procedure can be carried out between the fax adaptors and consequently end‑to‑end. Once the connection is established, both circuit 106 and circuit 109 are clamped to the ON condition by the fax adaptor at the IWF, so fixing a full duplex mode throughout the whole facsimile phase of the call. |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 8.2.2.3 Manual calling | Refer to diagram in figure II.5/GSM 03.46. When the call is answered, the RLP will be established across the radio interface providing circuit 108/2 in ON condition.
In the case where a fax adaptor is used, the mobile facsimile terminal must be connected to line by manual intervention at this stage, and will cause the fax adaptor to turn on circuit 108/2 (according to CCITT Recommendation V.24) towards the MT.
In the case where a GSM facsimile machine is used, circuit 108/2 shall be turned on when the GSM facsimile machine is connected to line by manual intervention.
On completion of the synchronization process across the radio interface or RLP establishment, the modem at the IWF will be automatically selected and send CNG (mandatory) to PSTN facsimile terminal. Also circuit 107 shall be turned on by the MT.
In the case where a fax adaptor is used, the receipt of circuit 107 shall cause the fax adaptor to connect to line.
The analogue links at both the PSTN side and the mobile side (where a fax adaptor is used) will be established in accordance with the appropriate CCITT V. series recommendation.
Note that circuit 109 and circuit 106 at the R interface of the MT must be turned on by the fax adaptor at the IWF before any further procedure can be carried out between the fax adaptors and consequently end‑to‑end. Once the connection is established, both circuit 106 and circuit 109 are clamped to the ON condition by the fax adaptor at the IWF, so fixing a full duplex mode throughout the whole facsimile phase of the call. |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 9 Interworking to fixed networks | PSTN and ISDN only are considered, both used as transit networks to complement the PLMN in the end‑to‑end connection between facsimile group 3 terminal, figure 7/GSM 03.46.
I W F
┌ ─ ─ ─ ─ ─ ─ ─ ─ ─ ─ ─ ─ ┐
:┌────────┐┌─────────┐ :
:│Protocol││ V.21 │ :
┌─────┐ :│Control ││ ┌───────┴─┐ :
│ │ v v :│Monitor.│└─┤ V.27ter │ : ╔══════════╗2-w
╔╧═════╧╗ ╔═════════════╗╔════╗│ │╔═════════╗:│Trans- │ │ ┌───────┴─┐:───────────╢ PSTN ╟───
║ TE2 ╟──╢ Fax adaptor ╟╢ MT ╟┘ └╢ BSS/MSC ╟:│coding │ └─┤ V.29 │: ╚══════════╝
╚═══════╝ ╚═════════════╝╚════╝ ╚═════════╝:│Buffers │ │ Modem │:
/ \ :│ │ └─────────┘: ┌─────┐ ╔══════════╗
/ \ :│ │ :─┤Codec├─┼─╢ ISDN ╟───
/ \ :│L2RBOP │┌─────────────┐: └─────┘ : ╚══════════╝
/ \ :│Mapping ││Tone handling│: 3.1kHz
/ \ :└────────┘└─────────────┘: audio
┌ ─ ─ ─ ─ ─ ─ ─ ─ ─ ─ ─ ─ ┐ └ ─ ─ ─ ─ ─ ─ ─ ─ ─ ─ ─ ─ ┘
:┌─────────┐ ┌────────┐:
:│ V.21 │ │Protocol│:
:│ ┌───────┴─┐ │Control │:
:└─┤ V.27ter │ │Monitor.│:
: │ ┌───────┴─┐│Trans- │:
2-w: └─┤ V.29 ││coding │:
────: │ Modem ││Buffers │:
: └─────────┘│LAPB- │:
: │Handling│:
:┌─────────────┐│L2RBOP │:
:│Tone handling││Mapping │:
:└─────────────┘└────────┘:
└ ─ ─ ─ ─ ─ ─ ─ ─ ─ ─ ─ ─ ┘
Figure 7/03.46: Network interworking |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 9.1 Interworking to PSTN | As the standard access of facsimile group 3 terminals for this Teleservice is a 2‑wire analogue interface, all the technical requirements for network interworking to PSTN are identical in principle to those encountered for the terminal connection to the MT. The key functional block is the fax adaptor described in clause 6 of the present document.
As far as network interworking is concerned, the main function to be performed by such a block is the correct managing of a composite modem, in accordance with the requirements of CCITT Recommendation T.30:
‑ CCITT Recommendation V.21 synchronous mode, as standard facility for all BCS phases;
‑ CCITT Recommendation V.27ter for message speeds of 4 800 and 2 400 bit/s;
‑ CCITT Recommendation V.29 for message speeds of 9 600 and 7 200 bit/s.
The mechanism for selecting the right modem is the following:
‑ the actual message speed is obtained by detecting the DCS frame (see table 2/CCITT Recommendation T.30) while in BCS phase;
‑ on entering the message phase, there is an interchange between the V.21 modem and the actual modem agreed upon between the terminals for message transmission;
‑ on exiting the message phase (RTC) the CCITT Recommendation V.21 modem is selected again.
Times for settling the modem will be in accordance with the requirements of CCITT Recommendation T.30. |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 9.2 Interworking to ISDN | The use of 3.1 kHz audio bearer capability of ISDN allows for an interworking of PLMN very similar in practice to the scheme for PSTN, figure 7/GSM 03.46. The fax adaptor function is in conformance with the description given in clause 4 and subclause 7.1 of the present document.
Annex A (normative):
Structure and contents of the fax adaptor protocol elements |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 1 Principle structure of an element | Each FA protocol element consists of the element discriminator (one single octet) and the optional information field (arbitrary length).
The elements are transmitted with octet 0, bit 1 first. Received information is forwarded with the same bit sequence as received.
octet: 0 1 ..... n
┌───────────────┬────────────────────────────────────────────┐
│ Element │ CCITT Recommendation T.30 Protocol Element │
│ Discriminator │ (optional) │
└───────────────┴────────────────────────────────────────────┘
│<-- 1 octet -->│<--------------- n octets ----------------->│
Figure A.1/03.46: Principle FA protocol element structure |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 2 Element discriminator coding | |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 2.1 BCS element | bit 8 7 6 5 4 3 2 1
┌───┬───┬───┬───┬───┬───┬───┬───┐
│ X │ 0 │ y │ y │ 0 │ 0 │ 0 │ 0 │
└───┴───┴───┴───┴───┴───┴───┴───┘
│ └──┬──┘
│ 0 1 = begin of a BCS frame
│ 1 0 = end of a BCS frame
│ 0 0 = middle of a BCS frame
│ 1 1 = entire BCS frame
0 = non-final frame
1 = final frame
Figure A.2/03.46: Element discriminator of a BCS element |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 2.2 BCS abort element | bit 8 7 6 5 4 3 2 1
┌───┬───┬───┬───┬───┬───┬───┬───┐
│ 1 │ 0 │ 1 │ 0 │ 0 │ 0 │ 0 │ 1 │
└───┴───┴───┴───┴───┴───┴───┴───┘
Figure A.3/03.46: Element discriminator of a BCS abort element |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 2.3 BCS transmit request element | bit 8 7 6 5 4 3 2 1
┌───┬───┬───┬───┬───┬───┬───┬───┐
│ 0 │ 1 │ 0 │ 0 │ 0 │ 0 │ 0 │ 1 │
└───┴───┴───┴───┴───┴───┴───┴───┘
Figure A.4/03.46: Element discriminator of a BCS transmit request element |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 2.4 Preamble element | bit 8 7 6 5 4 3 2 1
┌───┬───┬───┬───┬───┬───┬───┬───┐
│ 0 │ 1 │ 0 │ 0 │ 0 │ 0 │ 0 │ 0 │
└───┴───┴───┴───┴───┴───┴───┴───┘
Figure A.5/03.46: Element discriminator of a preamble element |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 2.5 Normal fax data element | bit 8 7 6 5 4 3 2 1
┌───┬───┬───┬───┬───┬───┬───┬───┐
│ 0 │ 1 │ 0 │ 0 │ 1 │ 0 │ 0 │ 0 │
└───┴───┴───┴───┴───┴───┴───┴───┘
Figure A.6/03.46: Element discriminator of a normal fax data element |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 2.6 Error correction fax data element | bit 8 7 6 5 4 3 2 1
┌───┬───┬───┬───┬───┬───┬───┬───┐
│ 0 │ 1 │ 0 │ 0 │ 1 │ 0 │ 0 │ 1 │
└───┴───┴───┴───┴───┴───┴───┴───┘
Figure A.7/03.46: Element discriminator of an error correction fax data element |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 2.7 End of data element | bit 8 7 6 5 4 3 2 1
┌───┬───┬───┬───┬───┬───┬───┬───┐
│ 0 │ 1 │ 0 │ 0 │ 1 │ 0 │ 1 │ 0 │
└───┴───┴───┴───┴───┴───┴───┴───┘
Figure A.8/03.46: Element discriminator of an end of data element |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 2.8 TCF element | bit 8 7 6 5 4 3 2 1
┌───┬───┬───┬───┬───┬───┬───┬───┐
│ 1 │ 0 │ 0 │ 1 │ 0 │ 0 │ 0 │ 1 │
└───┴───┴───┴───┴───┴───┴───┴───┘
Figure A.9/03.46: Element discriminator of a TCF element |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 3 Information field content | |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 3.1 BCS element | CCITT Recommendation ╔════════════════════════╗
T.30 frame ║ FCF + [FIF] ║
╚════════════════════════╝
| |
FA protocol ╔═══╤════╤════════════════════════╗
element ║ D │ SN │ ║
╚═══╧════╧════════════════════════╝
D = discriminator octet
SN = sequence number (0 .. 255), bit 1 = LSB
Figure A.10/03.46: Information field content of a BCS element |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 3.2 BCS abort element | no information field available |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 3.3 BCS transmit request element | FA protocol ╔═══╤════╗
element ║ D │ SN ║
╚═══╧════╝
D = discriminator octet
SN = sequence number (0 .. 255),
bit 1 = LSB = Least significant bit |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 3.4 Preamble element | no information field available |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 3.5 Normal fax data element | transcoded ═╤═══════════════════════════════════════╤═
facsimile │ max. 936 bits of facsimile coded data │
data ═╧═══════════════════════════════════════╧═
│ │
FA protocol ╔═══╤═══════════════════════════════════════╗
element ║ D │ ║
╚═══╧═══════════════════════════════════════╝
D = discriminator octet
Figure A.11/03.46: Information field content of a normal fax data element |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 3.6 Error correction fax data element | CCITT Recommendation
T.4 frame ╔════════════════════════╗
(FCD or RCP) ║ FCF + [FIF] ║
╚════════════════════════╝
| |
FA protocol ╔═══╤════════════════════════╗
element ║ D │ ║
╚═══╧════════════════════════╝
D = discriminator octet
Figure A.12/03.46: Information field content of an error correction fax data element |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 3.7 End of data element | no information field available |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 3.8 TCF element | bit 8 7 6 5 4 3 2 1
┌───┬───┬───┬───┬───┬───┬───┬───┐
│ 0 │ 0 │ 0 │ 0 │ 0 │ 0 │ 0 │ X │
└───┴───┴───┴───┴───┴───┴───┴───┘
X = 0 : TCF_OK
X = 1 : TCF_NOK
Figure A.13/03.46: Information field content of a TCF element |
1b3a41393e6b9c2d9c5a8ec454e96ca8 | 03.46 | 4 Relationship of FA protocol elements with LAPB, L2RBOP and RLP | Refer also to GSM 07.03
FA protocol ╔═╤════════════════════════╗
element ║D│ optional information ║
╚═╧════════════════════════╝
| |
LAPB I-frame ╔═╤═╤══════════════════════════╤═══╗
information ║A│C│ │FCS║
field ╚═╧═╧══════════════════════════╧═══╝
| |
╔══╤══════════════════════════╤══╤══╤═════════╗
L2RBOP PDU ║S1│ │S2│S3│arbitrary║
╚══╧══════════════════════════╧══╧══╧═════════╝
| |
RLP ╔═╤═════════════════════════════════════════════╤═╗
block ║H│ │F║
╚═╧═════════════════════════════════════════════╧═╝
D = Discriminator octet,
A = address field, C = control field,
S1, S2, S3 = status octets,
H = RLP header, F = RLP FCS
Figure A.14/03.46: Mapping for a "short" FA protocol element
FA protocol ╔═╤═══════════════════ ═════════╗
element ║D│ information ... ║
╚═╧═══════════════════ ═════════╝
| |
LAPB I-frame ╔═╤═╤═════════════════════ ═════════╤═══╗
information ║A│C│ │ │ ... │ │FCS║
field ╚═╧═╧═════════════════════ ═════════╧═══╝
/ / \ \ \ \
╔══╤════════╗╔══╤════════╗ ╔══╤══════╤══╤══╗
L2RBOP PDUs ║S0│ #1 ║║S0│ #2 ║ ... ║S1│ #n │S2│S3║
╚══╧════════╝╚══╧════════╝ ╚══╧══════╧══╧══╝
/ / \ \ \ \
RLP ╔═╤════════════╤═╗╔═╤════════════╤═╗ ╔═╤═════════════╤═╗
blocks ║H│ │F║║H│ │F║.║H│ │F║
╚═╧════════════╧═╝╚═╧════════════╧═╝ ╚═╧═════════════╧═╝
D = Discriminator octet,
A = address field, C = control field,
S0, S1, S2, S3 = status octets,
H = RLP header, F = RLP FCS
Figure A.15/03.46: Mapping for a "long" FA protocol element
Appendix I (informative): Abbreviations from CCITT Recommendation T.30 and T.4
Table I.1/03.46: Abbreviations from CCITT Recommendation T.30
Abbre‑
Function
Signal format
T.30
T.30
viation
standard
err.corr.
CED
Called station identification
2100 Hz
X
X
CFR
Confirmation to receive
X010 0001
X
X
CRP
Command repeat
X101 1000
X
X
CIG
Calling subscriber identification
1000 1000
X
X
CNG
Calling tone
1100 Hz
X
X
CSI
Called subscriber identification
0000 0010
X
X
CTC
Continue to correct
X100 1000
X
CTR
Response to continue to correct
X010 0011
X
DCN
Disconnect
X101 1111
X
X
DCS
Digital command signal
X100 0001
X
X
DIS
Digital identification signal
0000 0001
X
X
DTC
Digital transmit command
1000 0001
X
X
EOM
End of message
X111 0001
X
EOP
End of procedure
X111 0100
X
EOR
End of retransmission
X111 0011
X
ERR
Response to end of retransmission
X011 1000
X
FCD
Facsimile coded data
0110 0000
X
FCF
Facsimile control field
‑‑‑
X
X
FCS
Frame checking sequence
16 bits
X
X
FIF
Facsimile information field
‑‑‑
X
X
FTT
Failure to train
X010 0010
X
X
MCF
Message confirmation
X011 0001
X
X
MPS
Multi‑page signal
X111 0010
X
NSC
Non‑standard facilities command
1000 0100
X
X
NSF
Non‑standard facilities
0000 0100
X
X
NSS
Non‑standard set‑up
X100 0100
X
X
PIN
Procedural interrupt negative
X011 0100
X
X
PIP
Procedural interrupt positive
X011 0101
X
X
PIS
Procedure interrupt signal
462 Hz
X
X
PPR
Partial page request
X011 1101
X
PPS
Partial page signal
X111 1101
X
PRI
Procedure interrupt
X111 XXXX
X
RCP
Return to control for partial page
0110 0001
X
RNR
Receive not ready
X011 0111
X
RR
Receive ready
X111 0110
X
RTN
Retrain negative
X011 0010
X
X
RTP
Retrain positive
X011 0011
X
X
TCF
Training check frame
0... 1.5s
X
X
TSI
Transmitting subscriber identification
X100 0010
X
X
Table I.2/03.46: Abbreviations from CCITT Recommendation T.4
Abbre‑
Function
Signal format
viation
EOL
End of line
0000 0000 0001
RTC
Return to control
6 * EOL
Appendix II (informative): Procedure examples
106, 107, 108/2, 109: circuits according to CCITT Recommendation V.24
(1) manual intervention
(2) mandatory
(3) locally generated by the fax adaptor at IWF
(4) optionally
(5) triggered by delayed CT108.2 (3 sec)
Figure II.1a/03.46: Mobile terminated call ‑ speech then facsimile DCD mobile terminated
106, 107, 108/2, 109: circuits according to CCITT Recommendation V.24
(1) manual intervention
(2) mandatory
(3) locally generated by fax adaptor at IWF
(4) optionally
(5) triggered by delayed CT108.2 (3 sec)
(6) transmitted only if neither CED nor BCS is already received
Figure II.1b/03.46: Mobile terminated call ‑ speech then facsimile DCD mobile originated
106, 107, 108/2, 109, 125: circuits according to CCITT Recommendation V.24
(1) manual or automatic operation
(2) mandatory
(3) either after synchronization or RLP establishment
(4) locally generated by fax adaptor at IWF
Figure II.2/03.46: Mobile terminated call ‑ auto answer
106, 107, 108/2, 109: circuits according to CCITT Recommendation V.24
(1) manual intervention
(2) mandatory
(3) locally generated by fax adaptor at IWF
(4) optionally
(5) triggered by delayed CT108.2 (3 sec)
Figure II.3a/03.46: Mobile originated call ‑ speech then facsimile DCD mobile terminated
106, 107, 108/2, 109: circuits according to CCITT Recommendation V.24
(1) manual intervention
(2) mandatory
(3) locally generated by fax adaptor at IWF
(4) optionally
(5) triggered by delayed CT108.2 (3 sec)
(6) transmitted only if neither CED nor BCS is already received
Figure II.3b/03.46: Mobile originated call ‑ speech then facsimile DCD mobile originated
106, 107, 108/2, 109: circuits according to CCITT Recommendation V.24
(1) manual intervention
(2) mandatory
(3) PSTN fax terminal may be manually or automatically answered
(4) either after synchronization or RLP establishment
(5) locally generated by fax adaptor at IWF
Figure II.4/03.46: Mobile originated call ‑ auto calling
106, 107, 108/2, 109: circuits according to CCITT Recommendation V.24
(1) manual intervention
(2) mandatory
(3) PSTN fax terminal may be manually or automatically answered
(4) either after synchronization or RLP establishment
(5) locally generated by fax adaptor at IWF
Figure II.5/03.46: Mobile originated call ‑ manual calling
Figure II.6/03.46: Mobile originated facsimile transmission
Figure II.7/03.46: Mobile terminated facsimile transmission
Figure II.8/03.46: Mobile originated facsimile transmission (error correction mode)
Figure II.9/03.46: Mobile originated facsimile transmission ‑ error recovery (example)
Figure II.10/03.46: Mobile terminated facsimile transmission ‑ error recovery (example)
Figure II.11/03.46: Mobile originated facsimile transmission ‑ error recovery (example)
Figure II.12/03.46: Mobile originated facsimile transmission ‑ error recovery (example)
Figure II.13/03.46: Mobile originated facsimile transmission ‑ error recovery (example)
Annex B (informative):
Change Request History
Change history
SMG No.
TDoc. No.
CR. No.
Section affected
New version
Subject/Comments
SMG#11
4.1.2
ETSI Publication
SMG#20
5.0.0
Release 1996 version
SMG#27
6.0.0
Release 1997 version
SMG#29
7.0.0
Release 1998version
TSG#06
8.0.0
Agreed to be created as a version 8 for Release 1999
History
Document history
V7.0.0
August 1999
Publication
V8.0.0
January 2000 |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 1 Scope | The present document specifies the digital test sequences for the GSM half rate speech codec. These sequences test for a bit exact implementation of the half rate speech transcoder (GSM 06.20 [2]), Voice Activity Detector (GSM 06.42 [6]), comfort noise (GSM 06.22 [4]) and the discontinuous transmission (GSM 06.41 [5]). |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 2 References | The following documents contain provisions which, through reference in this text, constitute provisions of the present document.
• References are either specific (identified by date of publication, edition number, version number, etc.) or non‑specific.
• For a specific reference, subsequent revisions do not apply.
• For a non-specific reference, the latest version applies.
• A non-specific reference to an ETS shall also be taken to refer to later versions published as an EN with the same number.
• For this Release 1999 document, references to GSM documents are for Release 1999 versions (version 8.x.y).
[1] GSM 01.04: "Digital cellular telecommunications system (Phase 2+); Abbreviations and acronyms".
[2] GSM 06.20: "Digital cellular telecommunications system (Phase 2+); Half rate speech; Half rate speech transcoding".
[3] GSM 06.21: "Digital cellular telecommunications system (Phase 2+); Half rate speech; Substitution and muting of lost frame for half rate speech traffic channels".
[4] GSM 06.22: "Digital cellular telecommunications system (Phase 2+); Half rate speech; Comfort noise aspects for half rate speech traffic channels".
[5] GSM 06.41: "Digital cellular telecommunications system (Phase 2+); Half rate speech; Discontinuous Transmission (DTX) for half rate speech traffic channels".
[6] GSM 06.42: "Digital cellular telecommunications system (Phase 2+); Half rate speech; Voice Activity Detector (VAD) for half rate speech traffic channels".
[7] GSM 06.06: "Digital cellular telecommunications system (Phase 2+); Half rate speech; ANSI‑C code for the GSM half rate speech codec".
[8] GSM 06.02: "Digital cellular telecommunications system (Phase 2+); Half rate speech; Half rate speech coding functions". |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 3 Definitions and abbreviations | |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 3.1 Definitions | Definition of terms used in the present document can be found in GSM 06.20 [2], GSM 06.21 [3], GSM 06.22 [4], GSM 06.41 [5] and GSM 06.42 [6]. |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 3.2 Abbreviations | For the purposes of the present document, the following abbreviations apply:
ETS European Telecommunication Standard
GSM Global System for Mobile communications
For abbreviations not given in this clause, see GSM 01.04 [1]. |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 4 General | Digital test sequences are necessary to test for a bit exact implementation of the half rate speech transcoder (GSM 06.20 [2]), Voice Activity Detector (GSM 06.42 [6]), comfort noise (GSM 06.22 [4]) and the discontinuous transmission (GSM 06.41 [5]).
The test sequences may also be used to verify installations of the ANSI C code in GSM 06.06 [7].
Clause 5 describes the format of the files which contain the digital test sequences. Clause 6 describes the test sequences for the speech transcoder. Clause 7 describes the test sequences for the VAD, comfort noise and discontinuous transmission.
Clause 8 describes the method by which synchronization is obtained between the test sequences and the speech codec under test.
Clause 9 describes the optional acceptance testing of the speech encoder and decoder in the TRAU by means of 8 bit A- or -law compressed test sequences on the A-Interface.
Electronic copies of the digital test sequences are provided as clause 10, these digital test sequences are contained in archive en_300968v080001p0.ZIP which accompanies the present document. |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 5 Test sequence format | This clause provides information on the format of the digital test sequences for the GSM half rate speech transcoder (GSM 06.20 [2]), Voice Activity Detector (GSM 06.42 [6]), comfort noise (GSM 06.22 [4]) and the discontinuous transmission (GSM 06.41 [5]). |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 5.1 File format | The test sequence files are provided in archive en_300968v080001p0.ZIP which accompanies the present document.
Following decompression, by execution of the 11 "disk*.exe" files, four types of file are provided:
‑ Files for input to the GSM half rate speech encoder: *.INP
‑ Files for comparison with the encoder output: *.COD
‑ Files for input to the GSM half rate speech decoder: *.DEC
‑ Files for comparison with the decoder output: *.OUT
Tables 1, 2, 3 and 4 define the formats of the four types of file. Each parameter in these tables is contained in a 16 bit word except for the samples of the 8 bit PCM test sequences, which are contained in an 8 bit word each. The left or right justification is indicated in the tables. The size and location of speech parameters in the encoder output (*.COD) and decoder input files (*.DEC) are described in GSM 06.20 [2]. |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 5.2 Codec homing | Each *.INP file includes two homing frames at the start of the test sequence. The function of these frames is to reset the speech encoder state variables to their initial value. In the case of a correct installation of the ANSI‑C simulation (GSM 06.06 [7]), all speech encoder output frames shall be identical to the corresponding frame in the *.COD file. In the case of a correct hardware implementation undergoing type approval, the first speech encoder output frame is undefined and need not be identical to the first frame in the *.COD file, but all remaining speech encoder output frames shall be identical to the corresponding frames in the *.COD file.
Each *.DEC file includes two homing frames at the start of the test sequence. The function of these frames is to reset the speech decoder state variables to their initial value. In the case of a correct installation of the ANSI‑C simulation (GSM 06.06 [7]), all speech decoder output frames shall be identical to the corresponding frame in the *.OUT file. In the case of a correct hardware implementation undergoing type approval, the first speech decoder output frame is undefined and need not be identical to first frame in the *.OUT file, but all remaining speech decoder output frames shall be identical to the corresponding frames in the *.OUT file.
Table 1: Encoder input sequence (*.INP) format
Name
Description
No. of bits
Justification
s(n)
Encoder input signal
13
Left
Table 2: Encoder output sequence (*.COD) format
Name
Description
No. of words
Justification
Speech
Speech parameters to the channel encoder
18
Right
Additional information
VAD
SP
Voice activity detection flag
SP flag
1
1
Right
Right
Table 3: Decoder input sequence (*.DEC) format
Name
Description
No. of bits/words
Justification
Speech parameters
Speech parameters to the channel encoder
18 words
Right
BFI flag
UFI flag
SID flag
TAF flag
Bad Frame Indicator
Unreliable Frame Indicator
SIlence Descriptor
Time Alignment Flag
1 bit / 1 word
1 bit / 1 word
2 bits / 1 word
1 bit / 1 word
Right
Right
Right
Right
Table 4: Decoder output sequence (*.OUT) format
Name
Description
No. of bits
Justification
s'(n)
Decoder output signal
13
Left |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 6 Speech codec test sequences | This clause describes the test sequences designed to exercise the GSM half rate speech transcoder (GSM 06.20 [2]). |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 6.1 Codec configuration | The speech encoder shall be configured to operate in the non‑DTX mode. The VAD and SP flags shall be set to 1 at the speech encoder output. |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 6.2 Speech codec test sequences | Table 5 lists the location and size of the speech codec test sequences. |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 6.2.1 Speech encoder test sequences | Three encoder input sequences are provided:
‑ SEQ01.INP ‑ Sequence for exercising the LPC vector quantization codebooks;
‑ SEQ02.INP ‑ Sequence for exercising the long term predictor codebooks;
‑ SEQ03.INP ‑ Sequence for exercising the remaining excitation codebooks.
The SEQ01.INP sequence causes the GSM half rate speech encoder to select every vector in the three reflection coefficient vector quantizers at least once. In a correct implementation, the resulting speech encoder output parameters shall be identical to those specified in the SEQ01.COD sequence.
The SEQ02.INP sequence causes the encoder to select at least once every quantization level in the eight bit table of long term filter lags for the first subframe, and every quantization level in the four bit delta lag quantizer for subframes 2, 3, and 4. In a correct implementation, the resulting speech encoder output parameters shall be identical to those specified in the SEQ02.COD sequence.
The SEQ03.INP sequence causes the encoder to select each of the quantization levels at least once for the remaining GSM half rate speech coder parameters: R0 (frame energy), the soft interpolation decision for the LPC coefficients, the four voicing modes, the gain vectors (GSP0) for each of the voicing modes, and the voiced and unvoiced VSELP codebooks. The only exception to this is that two GSP0 levels in the unvoiced mode are not selected. However, these levels are exercised in the GSM half rate speech decoder as described below. In a correct implementation, the resulting speech encoder output parameters shall be identical to those specified in the SEQ03.COD sequence. |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 6.2.2 Speech decoder test sequences | Four speech decoder input sequences are provided:
‑ SEQ01.DEC;
‑ SEQ02.DEC;
‑ SEQ03.DEC;
‑ SEQ04.DEC.
The SEQ01.DEC, SEQ02.DEC, and SEQ03.DEC sequences test the operation of the GSM half rate speech decoder in the absence of channel errors. They are derived from the corresponding SEQXX.INP sequences. In a correct implementation, the resulting speech decoder output shall be identical to the SEQ01.OUT, SEQ02.OUT, and SEQ03.OUT sequences, respectively. Together, these three sequences exercise every quantization level in every codebook in the decoder, with the exception of two GSP0 levels in the unvoiced mode.
The SEQ04.DEC sequence is designed to test the GSM half rate speech decoder under conditions which can result from channel errors. In particular, it is the decoding of LTP lags at the lag table boundaries, given delta lag codes which if incorrectly decoded would point outside the eight bit lag table, that is being tested. Also, the two remaining GSP0 levels in the unvoiced mode are exercised by this sequence. In a correct implementation, the resulting speech decoder output shall be identical to the SEQ04.OUT sequence. |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 6.2.3 Codec homing sequence | In addition to the test sequences described above, two homing sequences are provided to assist in codec type approval testing. SEQ05.INP contains one encoder‑homing‑frame. SEQ05.DEC contains one decoder‑homing‑frame. The use of these sequences is described in GSM 06.02 [8].
Table 5: Location and size of speech codec test sequences
Disk No.
File Name
No. of frames
Size (bytes)
1
1
2
2
SEQ01.INP
SEQ01.COD
SEQ01.DEC
SEQ01.OUT
2 359
754 880
94 360
103 796
754 880
1
1
2
2
SEQ02.INP
SEQ02.COD
SEQ02.DEC
SEQ02.OUT
781
249 920
31 240
34 364
249 920
1
1
2
2
SEQ03.INP
SEQ03.COD
SEQ03.DEC
SEQ03.OUT
413
132 160
16 520
18 172
132 160
2
2
SEQ04.DEC
SEQ04.OUT
76
3 344
24 320
1
2
SEQ05.INP
SEQ05.DEC
1
320
44 |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 7 DTX test sequences | This clause describes the test sequences designed to exercise the VAD algorithm (GSM 06.42 [6]), comfort noise (GSM 06.22 [4]) and discontinuous transmission (GSM 06.41 [5]). |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 7.1 Codec configuration | The VAD, comfort noise and discontinuous transmission shall be tested in conjunction with the speech encoder [2]). The speech encoder shall be configured to operate in the DTX mode defined in GSM 06.22 [4]. |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 7.2 DTX test sequences | Each DTX test sequence consists of four files:
‑ Files for input to the GSM half rate speech encoder: *.INP
‑ Files for comparison with the encoder output *.COD
‑ Files for input to the GSM half rate speech decoder: *.DEC
‑ Files for comparison with the decoder output: *.OUT
The *.DEC files are generated from the corresponding *.COD files.
In a correct implementation, the speech encoder parameters generated by the *.INP file shall be identical to those specified in the *.COD file; and the speech decoder output generated by the *.DEC file shall be identical to that specified in the *.OUT file.
Table 6 lists the DTX test sequences and their size in frames. |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 7.2.1 Predictor values computation | The computation of the predictor values described in GSM 06.42 [6] is not tested explicitly, since the results from the computation are tested many times via the spectral comparison and threshold adaptation tests. |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 7.2.2 Spectral comparison | The spectral comparison algorithm described in GSM 06.42 [6] is tested by the following test sequence:
‑ DTX01.* |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 7.2.3 Threshold adaptation | The threshold adaptation algorithm described in GSM 06.42 [6] is tested by the following test sequence:
‑ DTX02.* |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 7.2.4 Periodicity detection | The periodicity detection algorithm described in GSM 06.42 [6] is tested by the following test sequence:
‑ DTX03.* |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 7.2.5 Tone detection | The tone detection algorithm described in GSM 06.42 [6] is tested by the following test sequence:
‑ DTX04.* |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 7.2.6 Safety and initialization | This sequence checks the safety paths used to prevent zero values being passed to the norm function. It checks the functions described in the adaptive filtering and energy computation, and the prediction values computation given in GSM 06.42 [6]. This sequence also checks the initialization of thvad and the rvad array:
‑ DTX05.* |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 7.2.7 Comfort noise test sequence | The test sequences described in sub‑clauses 7.2.2 to 7.2.6 are designed to exercise the VAD described in GSM 06.42 [6] and the discontinuous transmission described in GSM 06.41 [5]. The following test sequence is defined to exercise the comfort noise algorithm described in GSM 06.22 [4]:
‑ DTX06.* |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 7.2.8 Real speech and tones | The test sequences cannot be guaranteed to find every possible error. There is therefore a small possibility that an incorrect implementation produces the correct output for the test sequences, but fails with real signals. Consequently, an extra sequence is included, which consists of very clean speech, barely detectable speech and a swept frequency tone:
‑ DTX07.*
NOTE: Some of the DTX test sequences contain homing frames. The DTX test sequences are therefore only suitable for testing a single transcoding.
Table 6: Location and size of DTX test sequences
size
(bytes)
Disk No.
File Name
No. of Frames
*.INP
*.COD
*.DEC
*.OUT
3
DTX01
460
147 200
18 400
20 240
147 200
3
DTX02
886
283 520
35 440
38 984
283 520
3
DTX03
125
40 000
5 000
5 500
40 000
3
DTX04
317
101 440
12 680
13 948
101 440
3
DTX05
37
11 840
1 480
1 628
11 840
4
DTX06
240
76 800
9 600
10 560
76 800
4
DTX07
1 188
380 160
47 520
52 272
380 160 |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 8 Sequences for finding the 20 ms framing of the GSM half rate speech encoder | When testing the decoder, alignment of the test sequences used to the decoder framing is achieved by the air interface (testing of MS) or can be reached easily on the Abis‑interface (testing on network side).
When testing the encoder, usually there is no information available about where the encoder starts its 20 ms segments of speech input to the encoder.
In the following, a procedure is described to find the 20 ms framing of the encoder using special synchronization sequences. This procedure can be used for MS as well as for network side.
Synchronization can be achieved in two steps. First, bit synchronization has to be found. In a second step, frame synchronization can be determined. This procedure takes advantage of the codec homing feature of the half rate codec, which puts the codec in a defined home state after the reception of the first homing frame. On the reception of further homing frames, the output of the codec is predefined and can be triggered to. |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 8.1 Bit synchronization | The input to the speech encoder is a series of 13 bit long words (104 kbits/s, 13 bit linear PCM). When starting to test the speech encoder, no knowledge is available on bit synchronization, i.e. where the encoder expects its least significant bits, and where it expects the most significant bits.
The encoder homing frame consists of 160 samples, all set to zero with the exception of the least significant bit, which is set to one (0 0000 0000 0001 binary, or 0x0008 hex if written into 16 bit words left justified). If two such encoder homing frames are input to the encoder consecutively, the decoder homing frame is expected at the output as a reaction of the second encoder homing frame.
Since there are only 13 possibilities for bit synchronization, after a maximum of 13 trials bit synchronization can be reached. In each trial, three consecutive encoder homing frames are input to the encoder. If the decoder homing frame is not detected at the output, the relative bit position of the three input frames is shifted by one and another trial is performed. As soon as the decoder homing frame is detected at the output, bit synchronization is found, and the first step can be terminated.
The reason why three consecutive encoder homing frames are needed is that frame synchronization is not known at this stage. To be sure that the encoder reads two complete homing frames, three frames have to be input. Wherever the encoder has its 20 ms segmentation, it will always read at least two complete encoder homing frames.
An example of the 13 different frame triplets is given in sequence BITSYNC.INP (see table 7). |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 8.2 Frame synchronization | Once bit synchronization is found, frame synchronization can be found by inputting one special frame that delivers 160 different output frames, depending on the 160 different positions that this frame can possibly have with respect to the encoder framing.
This special synchronization frame was found by taking one input frame and shifting it through the positions 0 to 159. The corresponding 160 encoded speech frames were calculated and it was verified that all 160 output frames were different. When shifting the input synchronization frame, the samples at the beginning were set to 0x0008 hex, which corresponds to the samples of the encoder homing frame.
Before inputting this special synchronization frame to the encoder, again the encoder has to be reset by one encoder homing frame. A second encoder homing frame is needed to provoke a decoder homing frame at the output that can be triggered to. And since the framing of the encoder is not known at that stage, three encoder homing frames have to precede the special synchronization frame to ensure that the encoder reads at least two homing frames, and at least one decoder homing frame is produced at the output, serving as a trigger for recording.
The special synchronization frame preceded by the three encoder homing frames are given in SEQSYNC.INP. The corresponding 160 different output frames are given in SYNC000.COD through SYNC159.COD. The three digit number in the filename indicates the number of samples by which the input was retarded with respect to the encoder framing. By a corresponding shift in the opposite direction, alignment with the encoder framing can be reached. |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 8.3 Formats and sizes of the synchronization sequences | BIT SYNC.INP:
This sequence consists of 13 frame triplets. It has the format of the speech encoder input test sequences (13 bit left justified with the three least significant bits set to zero).
The size of it is therefore:
SIZE (BITSYNC.INP) = 13 * 3 * 160 * 2 bytes = 12480 bytes.
SEQSYNC.INP:
This sequence consists of 3 encoder reset frames and the special synchronization frame. It has the format of the speech encoder input test sequences (13 bit left justified with the three least significant bits set to zero).
The size of it is therefore:
SIZE (SEQSYNC.INP) = 4 * 160 * 2 bytes = 1280 bytes.
SYNCXXX.COD:
These sequences consists of 1 encoder output frame each. They have the format of the speech encoder output test sequences (16 bit words right justified). The values of the VAD and SP flags are set to one in these files.
The size of them is therefore:
SIZE (SYNCXXX.COD) = (18 + 2) * 2 bytes = 40 bytes
Table 7 summarizes this information.
Table 7: Location, size and justification of synchronization sequences
Disk No.
Purpose of Sequence
Name of Sequence
No. of Frames
Size in Bytes
Justification
5
Bit Synchronization
BITSYNC.INP
39
1 2480
Left
5
Frame Synchronization (input)
SEQSYNC.INP
4
1 280
Left
5
Frame Synchronization (output)
SYNC000.COD
SYNC001.COD
SYNC002.COD
"
"
"
SYNC159.COD
1
1
1
"
"
"
1
40
40
40
"
"
"
40
Right
Right
Right
"
"
"
Right |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 9 Trau Testing with 8 Bit A- and µ-law PCM Test Sequences | In the previous clauses tests for the transcoder in the TRAU are described using 13 bit linear test sequences. However, these 13 bit test sequences require a special interface in the Trau and do not allow testing in the field. In most cases the TRAU has to be set in special mode before testing.
As an option, the speech codec tests can be performed with A/ law compressed 8 bit PCM test sequences on the A interface. These modified input test sequences (*-X.INP) are generated from the original sequences by A or law compression. As an input to the encoder they result in modified encoder output sequences (*-X.COD). The same *.dec decoder input sequences as in clause 6.2.2. are then used to produce the output sequences *-X.OUT, which are A- or µ-law compressed.
The A- and µ-law compression and decompression does not change the homing frames at the encoder input. The format of all A- and -law PCM files *-X.INP and *-X.OUT is one sample (8 bit) per byte. The format of all other files is as described in clause 5.
All files are provided in archive en_300968v080001p0.ZIP which accompanies the present document. The 'X' in the tables below with the filenames stands for A (A-law) and U (-law), respectively. The decoder input files *.dec are the same as in Table 5 and are not described in this clause.
Table 8: Location and size of compressed 8 bit PCM speech codec test sequences
Disk No.
File Name
No. of frames
Size (bytes)
6/7
6/7
6/7
SEQ01-X.INP
SEQ01-X.COD
SEQ01-X.OUT
2 359
377 440
94 360
377 440
6/7
6/7
6/7
SEQ02-X.INP
SEQ02-X.COD
SEQ02-X.OUT
781
124 960
31 240
124 960
6/7
6/7
6/7
SEQ03-X.INP
SEQ03-X.COD
SEQ03-X.OUT
413
66 080
16 520
66 080
6/7
SEQ04-X.OUT
76
12 160
6/7
SEQ05-X.INP
1
160
Table 9: Location and size of compressed 8 bit PCM DTX test sequences
size
(bytes)
Disk No.
File Name
No. of Frames
*.INP
*.COD
*.OUT
8/9
DTX01-X
460
73 600
18 400
73 600
8/9
DTX02-X
886
141 760
35 440
141 760
8/9
DTX03-X
125
20 000
5 000
20 000
8/9
DTX04-X
317
50 720
12 680
50 720
8/9
DTX05-X
37
5 920
1 480
5 920
8/9
DTX06-X
240
38 400
9 600
38 400
8/9
DTX07-X
1 188
190 080
47 520
190 080
In addition to the testsequences above, special input (seqsyncX.inp) and output (syncxxxX.cod) sequences for frame synchronization are provided. The X again stands for A and law compressed PCM. The synchronization procedure is described in clause 8.
Table 10: Location, size and justification of compressed8 bit PCM test sequences
Disk No.
Purpose of Sequence
Name of Sequence
No. of Frames
Size in Bytes
Justification
10/11
Frame Synchronization (input)
SEQSYNCX.INP
4
640
-
10/11
Frame Synchronization (output)
SYNC000X.COD
SYNC001X.COD
SYNC002X.COD
"
"
"
SYNC159X.COD
1
1
1
"
"
"
1
40
40
40
"
"
"
40
Right
Right
Right
"
"
"
Right |
8b410df87e6e8f32e6bbafe6636ff843 | 06.07 | 10 Test sequences for the GSM half rate speech codec | NOTE: This clause is contained in archive en_300968v080001p0.ZIP which accompanies the present document.
Annex A (informative):
Change Request History
Change history
SMG No.
TDoc. No.
CR. No.
Section affected
New version
Subject/Comments
SMG#16
4.0.3
ETSI Publication
SMG#20
5.0.1
Release 1996 version
SMG#23
97-737
A003
5.1.1
UAP60 and Supplementary notes on 06.06 Call Graph Changes
SMG#27
6.0.0
Release 1997 version
SMG#28
6.0.1
ETSI Publication
SMG#29
7.0.0
Release 1998 version
7.0.1
Version update to 7.0.1 for Publication
SMG#31
8.0.0
Release 1999 version
8.0.1
Update to Version 8.0.1 for Publication
History
Document history
V8.0.0
July 2000
One-step Approval Procedure OAP 20001103: 2000-07-05 to 2000-11-03
V8.0.1
November 2000
Publication |
ad3845652ec99fdffd75badceec5143f | 06.32 | 0.1 Scope | The present document specifies the Voice Activity Detector (VAD) to be used in the Discontinuous Transmission (DTX) as described in GSM 06.31. It also specifies the test methods to be used to verify that a VAD complies with the technical specification.
The requirements are mandatory on any VAD to be used either in the GSM Mobile Stations (MS)s or Base Station Systems (BSS)s. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 0.2 References | The following documents contain provisions which, through reference in this text, constitute provisions of the present document.
• References are either specific (identified by date of publication, edition number, version number, etc.) or non‑specific.
• For a specific reference, subsequent revisions do not apply.
• For a non-specific reference, the latest version applies.
• A non-specific reference to an ETS shall also be taken to refer to later versions published as an EN with the same number.
• For this Release 1999 document, references to GSM documents are for Release 1999 versions (version 8.x.y).
[1] GSM 01.04: "Digital cellular telecommunications system (Phase 2+); Abbreviations and acronyms".
[2] GSM 06.10: "Digital cellular telecommunications system(Phase 2+); Full rate speech; Transcoding".
[3] GSM 06.12: "Digital cellular telecommunications system(Phase 2+); Full rate speech; Comfort noise aspect for full rate speech traffic channels".
[4] GSM 06.31: "Digital cellular telecommunications system(Phase 2+); Full rate speech; Discontinuous Transmission (DTX) for full rate speech traffic channels". |
ad3845652ec99fdffd75badceec5143f | 06.32 | 0.3 Abbreviations | Abbreviations used in the present document are listed in GSM 01.04 [1]. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 1 General | The function of the VAD is to indicate whether each 20 ms frame produced by the speech encoder contains speech or not. The output is a binary flag which is used by the TX DTX handler defined in GSM 06.31 [4].
The ETS is organized as follows.
Clause 2 describes the principles of operation of the VAD.
In clause 3, the computational details necessary for the fixed point implementation of the VAD algorithm are given. This clause uses the same notation as used for computational details in GSM 06.10.
The verification of the VAD is based on the use of digital test sequences. Clause 4 defines the input and output signals and the test configuration, whereas the detailed description of the test sequences is contained in clause A.2.
The performance of the VAD algorithm is characterized by the amount of audible speech clipping it introduces and the percentage activity it indicates. These characteristics for the VAD defined in the present document have been established by extensive testing under a wide range of operating conditions. The results are summarized in clause A.3. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2 Functional description | The purpose of this clause is to give the reader an understanding of the principles of operation of the VAD, whereas the detailed description is given in clause 3. In case of discrepancy between the two descriptions, the detailed description of clause 3 shall prevail.
In the following clauses of clause 2, a Pascal programming type of notation has been used to describe the algorithm. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.1 Overview and principles of operation | The function of the VAD is to distinguish between noise with speech present and noise without speech present. The biggest difficulty for detecting speech in a mobile environment is the very low speech/noise ratios which are often encountered. The accuracy of the VAD is improved by using filtering to increase the speech/noise ratio before the decision is made.
For a mobile environment, the worst speech/noise ratios are encountered in moving vehicles. It has been found that the noise is relatively stationary for quite long periods in a mobile environment. It is therefore possible to use an adaptive filter with coefficients obtained during noise, to remove much of the vehicle noise.
The VAD is basically an energy detector. The energy of the filtered signal is compared with a threshold; speech is indicated whenever the threshold is exceeded.
The noise encountered in mobile environments may be constantly changing in level. The spectrum of the noise can also change, and varies greatly over different vehicles. Because of these changes the VAD threshold and adaptive filter coefficients must be constantly adapted. To give reliable detection the threshold must be sufficiently above the noise level to avoid noise being identified as speech but not so far above it that low level parts of speech are identified as noise. The threshold and the adaptive filter coefficients are only updated when speech is not present. It is, of course, potentially dangerous for a VAD to update these values on the basis of its own decision. This adaptation therefore only occurs when the signal seems stationary in the frequency domain but does not have the pitch component inherent in voiced speech. A tone detector is also used to prevent adaptation during information tones.
A further mechanism is used to ensure that low level noise (which is often not stationary over long periods) is not detected as speech. Here, an additional fixed threshold is used.
A VAD hangover period is used to eliminate mid‑burst clipping of low level speech. Hangover is only added to speech‑bursts which exceed a certain duration to avoid extending noise spikes. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2 Algorithm description | The block diagram of the VAD algorithm is shown in figure 2.1. The individual blocks are described in the following clauses. ACF, N and sof are calculated in the speech encoder.
Figure 2.1: Functional block diagram of the VAD
The global variables shown in the block diagram are described as follows:
‑ ACF are auto‑correlation coefficients which are calculated in the speech encoder defined in GSM 06.10 (clause 3.1.4, see also clause A.1). The inputs to the speech encoder are 16 bit 2's complement numbers, as described in GSM 06.10, clause 4.2.0;
‑ av0 and av1 are averaged ACF vectors;
‑ rav1 are autocorrelated predictor values obtained from av1;
‑ rvad are the autocorrelated predictor values of the adaptive filter;
‑ N is the long term predictor lag value which is obtained every sub‑segment in the speech coder defined in GSM 06.10;
‑ ptch indicates whether the signal has a steady periodic component;
‑ sof is the offset compensated signal frame obtained in the speech coder defined in GSM 06.10;
‑ pvad is the energy in the current frame of the input signal after filtering;
‑ thvad is an adaptive threshold;
‑ stat indicates spectral stationarity;
‑ vvad indicates the VAD decision before hangover is added;
‑ vad is the final VAD decision with hangover included. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2.1 Adaptive filtering and energy computation | Pvad is computed as follows:
This corresponds to performing an 8th order block filtering on the input samples to the speech encoder, after zero offset compensation and pre‑emphasis. This is explained in clause A.1. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2.2 ACF averaging | Spectral characteristics of the input signal have to be obtained using blocks that are larger than one 20 ms frame. This is done by averaging the auto‑correlation values for several consecutive frames. This averaging is given by the following equations:
Where n represents the current frame, n‑1 represents the previous frame etc. The values of constants are given in table 2.1.
Table 2.1: Constants and variables for ACF averaging
Constant
Value
Variable
Initial value
frames
4
previous ACF's
av0 & av1
All set to 0 |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2.3 Predictor values computation | The filter predictor values aav1 are obtained from the auto‑correlation values av1 according to the equation:
where:
‑ ‑
R = | av1[0], av1[1], av1[2], av1[3], av1[4], av1[5], av1[6], av1[7] |
| av1[1], av1[0], av1[1], av1[2], av1[3], av1[4], av1[5], av1[6] |
| av1[2], av1[1], av1[0], av1[1], av1[2], av1[3], av1[4], av1[5] |
| av1[3], av1[2], av1[1], av1[0], av1[1], av1[2], av1[3], av1[4] |
| av1[4], av1[3], av1[2], av1[1], av1[0], av1[1], av1[2], av1[3] |
| av1[5], av1[4], av1[3], av1[2], av1[1], av1[0], av1[1], av1[2] |
| av1[6], av1[5], av1[4], av1[3], av1[2], av1[1], av1[0], av1[1] |
| av1[7], av1[6], av1[5], av1[4], av1[3], av1[2], av1[1], av1[0] |
‑ ‑
and:
‑ ‑ ‑ ‑
p = |av1[1]| a = |aav1[1]|
|av1[2]| |aav1[2]|
|av1[3]| |aav1[3]|
|av1[4]| |aav1[4]|
|av1[5]| |aav1[5]|
|av1[6]| |aav1[6]|
|av1[7]| |aav1[7]|
|av1[8]| |aav1[8]|
‑ ‑ ‑ ‑
aav1[0] = ‑1
av1 is used in preference to av0 as av0 may contain speech.
The autocorrelated predictor values rav1 are then obtained: |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2.4 Spectral comparison | The spectra represented by the autocorrelated predictor values rav1 and the averaged auto‑correlation values av0 are compared using the distortion measure dm defined below. This measure is used to produce a Boolean value stat every 20 ms, as given by these equations:
difference = |dm ‑ lastdm|
lastdm = dm
stat = difference < thresh
The values of constants and initial values are given in table 2.2.
Table 2.2: Constants and variables for spectral comparison
Constant
Value
Variable
Initial value
thresh
0.05
lastdm
0 |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2.5 Periodicity detection | The frequency spectrum of mobile noise is relatively stationary over quite long periods. The Inverse Filter Autocorrelated Predictor coefficients of the adaptive filter rvad are only updated when this stationarity is detected. Vowel sounds however, also have this stationarity, but can be excluded by detecting the periodicity of these sounds using the long term predictor lag values (Nj) which are obtained every sub‑segment from the speech codec defined in GSM 06.10. Consecutive lag values are compared. Cases in which one lag value is a factor of the other are catered for, however cases in which both lag values have a common factor, are not. This case is not important for speech input but this method of periodicity detection may fail for some sine waves. The Boolean variable ptch is updated every 20 ms and is true when periodicity is detected. It is calculated according to the following equation:
ptch = oldlagcount + veryoldlagcount >= nthresh
The following operations are done after the VAD decision and when the current LTP lag values (N0 .. N3) are available, this reduces the delay of the VAD decision. (N{‑1} = N3 of previous segment.)
lagcount = 0
for j = 0 to 3 do
begin
smallag = maximum(Nj,N{j‑1}) mod minimum(Nj,N{j‑1})
if minimum(smallag,minimum(Nj,N{j‑1})‑smallag) < lthresh
then increment(lagcount)
end
veryoldlagcount = oldlagcount
oldlagcount = lagcount
The values of constants and initial values are given in table 2..
Table 2.3: Constants and variables for periodicity detection
Constant
Value
Variable
Initial value
lthresh
nthresh
2
4
oldlagcount
veryoldlagcount
N3
0
0
40 |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2.6 Information tone detection | The tone flag is only evaluated in the downlink VAD. In the uplink VAD, tone detection is not performed and tone = false.
Computation of the tone flag is complex. It is therefore evaluated after the processing of the current speech encoder frame. In this way transmission of the speech or SID frame is not delayed.
Information tones and environmental noise can be classified by inspecting the short term prediction gain, information tones resulting in higher prediction gains than environmental noise. Tones can therefore be detected by comparing the prediction gain to a fixed threshold. By limiting the prediction gain calculation to a fourth order analysis, information signals consisting of one or two tones can be detected whilst minimizing the prediction gain for environmental noise.
The prediction gain decision is implemented by comparing the normalized prediction error with a threshold. This measure is used to evaluate the Boolean variable tone every 20 ms. The signal is classified as a tone if the prediction error is smaller than the threshold predth. This is equivalent to a prediction gain threshold of 13,5 dB.
Mobile noise can contain very strong resonances at low frequencies, resulting in a high prediction gain. A further test is therefore made to determine the pole frequency of a second order analysis of the signal frame. The signal is classified as noise if the frequency of the pole is less than 385 Hz. The pole frequency calculation is described in clause A.4.
The algorithm for detecting information tones is as follows:
tone = false
den = a[1]*a[1]
num = 4*a[2] ‑ a[1]*a[1]
if ( num <= 0 )
return
if (( a[1] < 0 ) AND ( num / den < freqth ))
return
4
prederr = MULT (1 ‑ RC[i]*RC[i])
i=1
if (prederr < predth)
tone = true
return
The values of the constants are given in table 2.4. The coefficients a[1..2] are transversal filter coefficients calculated from rc[1..2]. The calculation of the reflection coefficients rc[1..4] is described below.
The offset compensated signal frame sof[0..159] is multiplied by the Hanning window to give the windowed frame sofh[0..159]:
where
The auto‑correlation acfh[0..4] of the windowed signal frame is then calculated:
rc[1..4] are then calculated from acfh[0..4] using the Schur recursion described in the RPE‑LTP codec.
Table 2.4: Constants for information tone detection
Constant
Value
freqth
predth
0,0973
0,0158
NOTE: Reflection coefficients are available in the RPE‑LTP codec. However, they are calculated after pre‑emphasis using a rectangular window and do not give good tone detection results. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2.7 Threshold adaptation | A check is made every 20 ms to determine whether the VAD decision threshold (thvad) should be changed. This adaptation is carried out according to the flowchart shown in figure 2.2. The constants used are given in table 2.5.
Adaptation takes place in two different situations: firstly whenever ACF[0] is very low and secondly whenever there is a very high probability that speech and information tones are not present.
In the first case, the threshold is adapted if the energy of the input signal is less than pth. The threshold is set to plev without carrying out any further tests because at these very low levels the effect of the signal quantization makes it impossible to obtain reliable results from these tests.
In the second case, the decision threshold (thvad) and the adaptive filter coefficients (rvad) are only updated with the rav1 values when there is a very high probability that speech and information tones are not present. Adaptation occurs if the following conditions are met over a number (adp) of signal frames:
‑ stationarity is detected in the frequency domain;
‑ the signal does not contain a periodic component;
‑ information tones are not present.
The step‑size by which the threshold is adapted is not constant but a proportion of the current value (determined by constants dec and inc). The adaptation begins by experimentally multiplying the threshold by a factor of (1‑1/dec). If the new threshold is now higher than or equal to Pvad times fac then the threshold needed to be decreased and it is left at this new lower level. If, on the other hand, the new threshold level is less than Pvad times fac then the threshold either needed to be increased or kept constant. In this case it is set to Pvad times fac unless this would mean multiplying it by more than a factor of (1+1/inc) (in which case it is multiplied by a factor of (1+1/inc)). The threshold is never allowed to be greater than Pvad+margin.
Table 2.5: Constants and variables for threshold adaptation
Constant
Value
Variable
Initial value
pth
plev
fac
adp
inc
dec
margin
300 000
800 000
3.0
8
16
32
80 000 000
adaptcount
thvad
rvad[0]
rvad[1]
rvad[2]
rvad[3] to
rvad[8]
0
1 000 000
6
‑4
1
All 0
Figure 2.2: Flow diagram for threshold adaptation |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2.8 VAD decision | Prior to hangover the VAD decision condition is:
vvad = pvad > thvad |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2.9 VAD hangover addition | VAD hangover is only added to bursts of speech greater than or equal to burstconst blocks. The Boolean variable vad indicates the decision of the VAD with hangover included. The values of the constants are given in table 2.6. The hangover algorithm is as follows:
if vvad then increment(burstcount) else burstcount = 0
if burstcount >= burstconst then
begin
hangcount = hangconst;
burstcount = burstconst
end
vad = vvad or (hangcount >= 0)
if hangcount >= 0 then decrement(hangcount)
Table 2.6: Constants and variables for VAD hangover addition
Constant
Value
Variable
Initial value
burstconst
hangconst
3
5
burstcount
hangcount
0
‑1 |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3 Computational details | In the next paragraphs, the detailed description of the VAD algorithm follows the preceding high level description. This detailed description is divided in ten clauses related to the blocks of figure 2.1 (except periodicity updating) in the high level description of the VAD algorithm.
Those clauses are:
1) adaptive filtering and energy computation;
2) ACF averaging;
3) predictor values computation;
4) spectral comparison;
5) periodicity detection;
6) threshold adaptation;
7) VAD decision;
8) VAD hangover addition;
9) periodicity updating;
10) information tone detection.
The VAD algorithm takes as input the following variables of the RPE‑LTP encoder (see the detailed description of the RPE‑LTP encoder GSM 06.10):
‑ L_ACF[0..8], auto‑correlation function (GSM 06.10/4.2.4);
‑ scalauto, scaling factor to compute the L_ACF[0..8] (GSM 06.10/4.2.4);
‑ Nc, LTP lag (one for each sub‑segment, GSM 06.10/4.2.11);
‑ sof, offset compensated signal frame (GSM 06.10/4.2.2).
So four Nc values are needed for the VAD algorithm.
The VAD computation can start as soon as the L_ACF[0..8] and scalauto variables are known. This means that the VAD computation can take place after part 4.2.4 of GSM 06.10 (Auto‑correlation) of the LPC analysis clause of the RPE‑LTP encoder. This scheme will reduce the delay to yield the VAD information. The periodicity updating (included in clause 2.2.5) and information tone detection, are done after the processing of the current speech encoder frame.
All the arithmetic operations and names of the variables follow the RPE‑LTP detailed description. To increase the precision within the fixed point implementation, a pseudo‑floating point representation of some variables is used. This stands for the following variables (and related constants) of the VAD algorithm:
pvad: Energy of filtered signal;
thvad: Threshold of the VAD decision;
acf0: Energy of input signal.
For the representation of these variables, two integers (16 bits) are needed:
‑ one for the exponent (e_pvad, e_thvad, e_acf0);
‑ one for the mantissa (m_pvad, m_thvad, m_acf0).
The value e_pvad represents the lowest power of 2 just greater or equal to the actual value of pvad and the m_pvad value represents a integer which is always greater or equal to 16384 (normalized mantissa). It means that the pvad value is equal to:
This scheme guarantees a large dynamic range for the pvad value and always keeps a precision of 16 bits. All the comparisons are easy to make by comparing the exponents of two variables and the VAD algorithm needs only one pseudo‑floating point addition. All the computations related to the pseudo‑floating point variables require very simple 16 or 32 bits arithmetic operations defined in the detailed description of the RPE‑LTP encoder. This pseudo‑floating point arithmetic is only used in clauses 3.1 and 3.6.
Table 3.1 gives a list of all the variables of the VAD algorithm that must be initialized in the reset procedure and kept in memory for processing the subsequent frame of the RPE‑ LTP encoder. The types (16 or 32 bits) and initial values of all these variables are clearly indicated and their related clause is also mentioned. The bit exact implementation uses other temporary variables that are introduced in the detailed description whenever it is needed.
Table 3.1: Initial values for variables to be stored in memory
Names of variables:
type (# of bits):
Initialization:
Subclause:
Adaptive filter coefficients:
rvad[0]
16
24 576
3.1, 3.6
rvad[1]
16
‑16 384
3.1, 3.6
rvad[2]
16
4 096
3.1, 3.6
rvad[3..8]
16
0
3.1, 3.6
Scaling factor of ravd[0..8]:
normrvad
16
7
3.1, 3.6
Delay line of the auto‑correlation coefficients:
L_sacf[0..26]
32
0
3.2
L_sav0[0..35]
32
0
3.2
Pointers on the delay lines:
pt_sacf
16
0
3.2
pt_sav0
16
0
3.2
Distance measure:
L_lastdm
32
0
3.4
Periodicity counters:
oldlagcount
16
0
3.5, 3.9
veryoldlagcount
16
0
3.5, 3.9
Adaptive threshold:
e_thvad (exponent)
16
20
3.6
m_thvad (mantissa)
16
31 250
3.6
Counter for adaptation:
adaptcount
16
0
3.6
Hangover flags:
burstcount
16
0
3.8
hangcount
16
‑1
3.8
LTP lag memory:
oldlag
16
40
3.9
Tone Detection
tone
16
0
3.10 |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.1 Adaptive filtering and energy computation | This clause computes the e_pvad and m_pvad variables which represent the pvad value. It needs the L_ACF[0..8] and scalauto variables of the RPE‑LTP algorithm and the rvad[0..8] and normrvad variables produced by clause 3.6 of the VAD algorithm. It also computes a floating point representation of L_ACF[0] (e_acf0 and m_acf0) used in clause 3.6.
Test if L_ACF[0] is equal to 0:
IF ( scalauto < 0 ) THEN scalvad = 0;
ELSE scalvad = scalauto; / keep scalvad for use in clause 3.2 /
IF ( L_ACF[0] == 0 ) THEN
| e_pvad = ‑32768;
| m_pvad = 0;
| e_acf0 = ‑32768;
| m_acf0 = 0;
| EXIT /continue with clause 3.2/
Re‑normalization of the L_ACF[0..8]:
normacf = norm( L_ACF[0] );
| FOR i = 0 to 8:
| sacf[i] = ( L_ACF[i] << normacf ) >> 19;
| NEXT i:
Computation of e_acf0 and m_acf0:
e_acf0 = add( 32, (scalvad << 1 ) );
e_acf0 = sub( e_acf0, normacf);
m_acf0 = sacf[0] << 3;
Computation of e_pvad and m_pvad:
e_pvad = add( e_acf0, 14 );
e_pvad = sub( e_pvad, normrvad );
L_temp = 0;
| FOR i = 1 to 8:
| L_temp = L_add( L_temp, L_mult( sacf[i], rvad[i] ) );
| NEXT i:
L_temp = L_add( L_temp, L_mult( sacf[0], rvad[0] ) >> 1 );
IF ( L_temp <= 0 ) THEN L_temp = 1;
normprod = norm( L_temp );
e_pvad = sub( e_pvad, normprod );
m_pvad = ( L_temp << normprod ) >> 16; |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.2 ACF averaging | This clause uses the L_ACF[0..8] and the scalvad variables to compute the array L_av0[0..8] and L_av1[0..8] used in clause 3.3 and 3.4.
Computation of the scaling factor:
scal = sub( 10, (scalvad << 1) );
Computation of the arrays L_av0[0..8] and L_av1[0..8]:
| FOR i = 0 to 8:
| L_temp = L_ACF[i] >> scal;
| L_av0[i] = L_add( L_sacf[i], L_temp );
| L_av0[i] = L_add( L_sacf[i+9], L_av0[i] );
| L_av0[i] = L_add( L_sacf[i+18], L_av0[i] );
| L_sacf[ pt_sacf + i ] = L_temp;
| L_av1[i] = L_sav0[ pt_sav0 + i ];
| L_sav0[ pt_sav0 + i] = L_av0[i];
| NEXT i:
Update of the array pointers:
IF ( pt_sacf == 18 ) THEN pt_sacf = 0;
ELSE pt_sacf = add( pt_sacf, 9);
IF ( pt_sav0 == 27 ) THEN pt_sav0 = 0;
ELSE pt_sav0 = add( pt_sav0, 9); |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.3 Predictor values computation | This clause computes the array rav1[0..8] needed for the spectral comparison and the threshold adaptation. It uses the L_av1[0..8] computed in clause 3.2, and is divided in the three following clauses:
‑ Schur recursion to compute reflection coefficients.
‑ Step up procedure to obtain the aav1[0..8].
‑ Computation of the rav1[0..8]. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.3.1 Schur recursion to compute reflection coefficients | This clause is identical to the one used in the RPE‑LTP algorithm. The array vpar[1..8] is computed with the array L_av1[0..8] as an input.
Schur recursion with 16 bits arithmetic:
IF( L_av1[0] == 0 ) THEN
|== FOR i = 1 to 8:
| vpar[i] = 0;
|== NEXT i:
| EXIT; /continue with clause 3.3.2/
temp = norm( L_av1[0] );
|== FOR k=0 to 8:
| sacf[k] = ( L_av1[k] << temp ) >> 16;
|== NEXT k:
Initialize array P[..] and K[..] for the recursion:
|== FOR i=1 to 7:
| K[9‑i] = sacf[i];
|== NEXT i:
|== FOR i=0 to 8:
| P[i] = sacf[i];
|== NEXT i:
Compute reflection coefficients:
|== FOR n=1 to 8:
| IF( P[0] < abs( P[1] ) ) THEN
| |== FOR i = n to 8:
| | vpar[i] = 0;
| |== NEXT i:
| | EXIT; /continue with
| | clause 3.3.2/
| vpar[n] = div( abs( P[1] ), P[0] );
| IF ( P[1] > 0 ) THEN vpar[n] = sub( 0, vpar[n] );
| IF ( n == 8 ) THEN EXIT; /continue with clause 3.3.2/
|
| Schur recursion:
|
| P[0] = add( P[0], mult_r( P[1], vpar[n] ) );
|==== FOR m=1 to 8‑n:
| P[m] = add( P[m+1], mult_r( K[9‑m], vpar[n] ) );
| K[9‑m] = add( K[9‑m], mult_r( P[m+1], vpar[n] ) );
|==== NEXT m:
|
|== NEXT n: |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.3.2 Step‑up procedure to obtain the aav1[0..8] | Initialization of the step‑up recursion:
L_coef[0] = 16384 << 15;
L_coef[1] = vpar[1] << 14;
Loop on the LPC analysis order:
|= FOR m = 2 to 8:
|== FOR i = 1 to m‑1:
|== temp = L_coef[m‑i] >> 16; / takes the msb /
|== L_work[i] = L_add( L_coef[i], L_mult( vpar[m], temp ) );
|== NEXT i
|=
|== FOR i = 1 to m‑1:
|== L_coef[i] = L_work[i];
|== NEXT i
|=
|= L_coef[m] = vpar[m] << 14;
|= NEXT m:
Keep the aav1[0..8] on 13 bits for next clause:
| FOR i = 0 to 8:
| aav1[i] = L_coef[i] >> 19;
| NEXT i: |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.3.3 Computation of the rav1[0..8] | |= FOR i= 0 to 8:
|= L_work[i] = 0;
|== FOR k = 0 to 8‑i:
|== L_work[i] = L_add( L_work[i], L_mult( aav1[k], aav1[k+i] ) );
|== NEXT k:
|= NEXT i:
IF ( L_work[0] == 0 ) THEN normrav1 =0;
ELSE normrav1 = norm( L_work[0] );
|= FOR i= 0 to 8:
|= rav1[i] = ( L_work[i] << normrav1 ) >> 16;
|= NEXT i:
Keep the normrav1 for use in clause 3.4 and 3.6. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.4 Spectral comparison | This clause computes the variable stat needed for the threshold adaptation. It uses the array L_av0[0..8] computed in clause 3.2 and the array rav1[0..8] computed in clause 3.3.3.
Re‑normalize L_av0[0..8]:
IF ( L_av0[0] == 0 ) THEN
| FOR i = 0 to 8:
| sav0[i] = 4095;
| NEXT i:
ELSE
| shift = norm( L_av0[0] );
|= FOR i = 0 to 8:
|= sav0[i] = ( L_av0[i] << shift‑3 ) >> 16;
|= NEXT i:
Compute partial of dm:
L_ p = 0;
|= FOR i = 1 to 8:
|= L_ p = L_add( L_ p, L_mult( rav1[i], sav0[i] ) );
|= NEXT i:
Compute the division of partial by sav0[0]:
IF ( L_ p < 0 ) THEN L_temp = L_sub( 0, L_ p );
ELSE L_temp = L_ p;
IF ( L_temp == 0 ) THEN
| L_dm = 0;
| shift = 0;
ELSE
| sav0[0] = sav0[0] << 3;
| shift = norm( L_temp );
| temp = ( L_temp << shift ) >> 16;
| IF ( sav0[0] >= temp ) THEN
| | divshift = 0;
| | temp = div( temp, sav0[0] );
| ELSE
| | divshift = 1;
| | temp = sub( temp, sav0[0] );
| | temp = div( temp, sav0[0] );
|
| IF( divshift == 1 ) THEN L_dm = 32768;
| ELSE L_dm = 0;
|
| L_dm = L_add( L_dm, temp) << 1;
| IF( L_ p < 0 ) THEN L_dm = L_sub( 0, L_dm);
Re‑normalization and final computation of L_dm:
L_dm = ( L_dm << 14 );
L_dm = L_dm >> shift;
L_dm = L_add( L_dm, ( rav1[0] << 11 ) );
L_dm = L_dm >> normrav1;
Compute the difference and save L_dm:
L_temp = L_sub( L_dm, L_lastdm );
L_lastdm = L_dm;
IF ( L_temp < 0 ) THEN L_temp = L_sub( 0, L_temp );
L_temp = L_sub( L_temp, 3277 );
Evaluation of the stat flag:
IF ( L_temp < 0 ) THEN stat = 1;
ELSE stat = 0; |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.5 Periodicity detection | This clause just sets the ptch flag needed for the threshold adaptation.
temp = add( oldlagcount, veryoldlagcount );
IF ( temp >= 4 ) THEN ptch = 1;
ELSE ptch = 0; |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.6 Threshold adaptation | This clause uses the variables e_pvad, m_pvad, e_acf0 and m_acf0 computed in clause 3.1. It also uses the flags stat (see clause 3.4) and ptch (see clause 3.5). It follows the flowchart represented on figure 2.2.
Some constants, represented by a floating point format, are needed and a symbolic name (in capital letter) for their exponent and mantissa is used; table 3.2 lists all these constants with the symbolic names associated and their numerical constant values.
Table 3.2: List of constants
Constant
Exponent
Mantissa
pth
margin
plev
E_PTH = 19
E_MARGIN = 27
E_PLEV = 20
M_PTH = 18 750
M_MARGIN = 19 531
M_PLEV = 25 000
NOTE: Floating point representation of constants used in clause 3.6:
pth = 2(E_PTH)x(M_PTH/32768).
margin = 2(E_MARGIN)x(M_MARGIN/32768).
plev = 2(E_PLEV)x(M_PLEV/32768).
Test if acf0 < pth; if yes set thvad to plev:
comp = 0;
IF ( e_acf0 < E_PTH ) THEN comp = 1;
IF ( e_acf0 == E_PTH ) THEN IF ( m_acf0 < M_PTH ) THEN comp =1;
IF ( comp == 1 ) THEN
| e_thvad = E_PLEV;
| m_thvad = M_PLEV;
| EXIT; /continue with clause 3.7/
Test if an adaptation is needed:
comp = 0;
IF ( ptch == 1 ) THEN comp = 1;
IF ( stat == 0 ) THEN comp = 1;
IF ( tone == 1 ) THEN comp = 1;
IF ( comp == 1 ) THEN
| adaptcount = 0;
| EXIT; /continue with clause 3.7/
Incrementation of adaptcount:
adaptcount = add( adaptcount, 1 );
IF ( adaptcount <= 8 ) THEN EXIT; /continue with clause 3.7/
Computation of thvad‑(thvad/dec):
m_thvad = sub( m_thvad, (m_thvad >> 5 ) );
IF ( m_thvad < 16384) THEN
| m_thvad = m_thvad << 1;
| e_thvad = sub( e_thvad, 1 );
Computation of pvad*fac:
L_temp = L_add( m_pvad, m_pvad );
L_temp = L_add( L_temp, m_pvad );
L_temp = L_temp >> 1;
e_temp = add( e_pvad, 1 );
IF ( L_temp > 32767 ) THEN
| L_temp = L_temp >> 1;
| e_temp = add( e_temp, 1 );
m_temp = L_temp;
Test if thvad < pvad*fac:
comp = 0;
IF ( e_thvad < e_temp) THEN comp = 1;
IF (e_thvad == e_temp) THEN IF (m_thvad < m_temp) THEN comp =1;
Computation of minimum (thvad+(thvad/inc), pvad*fac) if comp = 1:
IF ( comp == 1 ) THEN
| Compute thvad +(thvad/inc).
| L_temp = L_add( m_thvad, (m_thvad >> 4 ) );
| IF ( L_temp > 32767 ) THEN
| | m_thvad = L_temp >> 1;
| | e_thvad = add( e_thvad,1 );
| ELSE m_thvad = L_temp;
| comp2 = 0;
| IF ( e_temp < e_thvad) THEN comp2 = 1;
| IF (e_temp == e__hvad) THEN IF (m_temp<m_thvad) THEN comp2 = 1;
| IF ( comp2 == 1 ) THEN
| | e_thvad = e_temp;
| | m_thvad = m_temp;
Computation of pvad + margin:
IF ( e_pvad == E_MARGIN ) THEN
| L_temp = L_add(m_pvad, M_MARGIN);
| m_temp = L_temp >> 1;
| e_temp = add( e_pvad, 1 );
ELSE
| IF ( e_pvad > E_MARGIN ) THEN
| | temp = sub( e_pvad, E_MARGIN );
| | temp = M_MARGIN >> temp;
| | L_temp = L_add( m_pvad, temp );
| | IF ( L_temp > 32767) THEN
| | | e_temp = add( e_pvad, 1 );
| | | m_temp = L_temp >> 1;
| | ELSE
| | | e_temp = e_pvad;
| | | m_temp = L_temp;
| ELSE
| | temp = sub( E_MARGIN, e_pvad );
| | temp = m_pvad >> temp;
| | L_temp = L_add( M_MARGIN, temp );
| | IF (L_temp > 32767) THEN
| | | e_temp = add( E_MARGIN, 1);
| | | m_temp = L_temp >> 1;
| | ELSE
| | | e_temp = E_MARGIN;
| | | m_temp = L_temp;
Test if thvad > pvad + margin:
comp = 0;
IF ( e_thvad > e_temp) THEN comp = 1;
IF (e_thvad == e_temp) THEN IF (m_thvad > m_temp) THEN comp =1;
IF ( comp == 1 ) THEN
| e_thvad = e_temp;
| m_thvad = m_temp;
Initialize new rvad[0..8] in memory:
normrvad = normrav1;
|= FOR i = 0 to 8:
|= rvad[i] = rav1[i];
|= NEXT i:
Set adaptcount to adp + 1:
adaptcount = 9; |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.7 VAD decision | This clause only outputs the result of the comparison between pvad and thvad using the pseudo‑floating point representation of thvad and pvad. The values e_pvad and m_pvad are computed in clause 3.1 and the values e_thvad and m_thvad are computed in clause 3.6.
vvad = 0;
IF (e_pvad > e_thvad) THEN vvad = 1;
IF (e_pvad == e_thvad) THEN IF (m_pvad > m_thvad) THEN vvad =1; |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.8 VAD hangover addition | This clause finally sets the vad decision for the current frame to be processed.
IF ( vvad == 1 ) THEN burstcount = add( burstcount, 1 );
ELSE burstcount = 0;
IF ( burstcount >= 3 ) THEN
| hangcount = 5;
| burstcount = 3;
vad = vvad;
IF ( hangcount >= 0 ) THEN
| vad = 1;
| hangcount = sub( hangcount, 1 ); |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.9 Periodicity updating | This clause must be delayed until the LTP lags are computed by the RPE‑LTP algorithm. The LTP lags called Nc in the speech encoder are renamed lags[0..3] (index 0 for the first sub‑ segment of the frame, 1 for the second and so on).
Loop on sub‑segments for the frame:
lagcount = 0;
|= FOR i = 0 to 3:
|= Search the maximum and minimum of consecutive lags.
|= IF ( oldlag > lags[i] ) THEN
|= | minlag = lags[i];
|= | maxlag = oldlag;
|= ELSE
|= | minlag = oldlag;
|= | maxlag = lags[i] ;
|=
|= Compute smallag (modulo operation not defined ):
|=
|= smallag = maxlag;
|== | FOR j = 0 to 2:
|== | IF (smallag >= minlag) THEN smallag =sub( smallag, minlag);
|== | NEXT j;
|=
|= Minimum of smallag and minlag ‑ smallag:
|=
|= temp = sub( minlag, smallag );
|= IF ( temp < smallag ) THEN smallag = temp;
|= IF ( smallag < 2 ) THEN lagcount = add( lagcount, 1 );
|= Save the current LTP lag.
|= oldlag = lags[i];
|= NEXT i:
Update the veryoldlagcount and oldlagcount:
veryoldlagcount = oldlagcount;
oldlagcount = lagcount; |
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