Javier Sanchez & Mamadou Tnioune Copyright 02007, ISTE Ltd.

Appendix 1

AMR Codec in UMTS

Originally developed to be used in GSM by the ETSI, the Adaptive Multi-Rate (AMR) speech codec [TS 26.071] was approved within the 3GPP forum in 1999 to be mandatory for circuit- and packet-switched speech in UMTS networks. An AMR speech codec adapts the error protection level to the local radio channel and traffic conditions so that it always selects the optimum channel and codec mode to deliver the best combination of speech quality and system capacity. AMR uses Multi-Rate Algebraic Code Excited Linear Prediction (MR-ACELP) scheme based on two different synthesis filters. It converts a narrowband speech signal (from 300 to 3,400 Hz) to 13-bit uniform Pulse Coded Modulated (PCM) samples with 8 kHz sample rate. This leads to 20 ms AMR frames consisting of 160 encoded speech samples. This means that the codec can switch mode, i.e. source bite rate, every 20 ms. AMR has 8 coded modes in UMTS systems, whereas in GSM AMR uses either 6 or 8 modes. The eight source rates vary from 4.75 to 12.2 kbps. It also contains a low rate encoding mode, called SIlence Descriptor (SID), which operates at 1.8 kbps to produce background noise and a non-transmission mode. The AMR codec dynamically adapts its error protection level to the channel error conditions. For instance, lower speech coding bit rate and more error protection schemes are used in bad channel conditions. This principle is illustrated in Figure A1.1 where AMR strives to change to the best curve associated to a given AMR mode. It has been shown that the degradation on the audio quality caused by a lower speech coding rate is compensated by increased robustness with the channel coding. Note, however, that this channel robustness is more beneficial in GSM than in UMTS due to the embedded fast power control used in WCDMA systems. Using a variable-rate transmission scheme also makes it possible to control the transmission power of the UE, a fact that is particularly useful when the UE

a noise frame. the frame is divided into a header. comfort noise parameters. The header contains the Frame Types and Frame Quality Indicator fields. or an empty frame. AMR frame Frame Type (4 bits) Frame Quality Indicator (1 bits) Mode Indication (3 bits) Mode Request (3 bits) Codec CRC bits (8 bits) Class A bits Class B bits Class C bits Core frame (speech or comfort noise) Header Auxiliary Information (for Mode Adaptation. AMR principle A1.1. i.376 UMTS suddenly attains its maximum transmit power: in CDMA: lower bit rates generally need lower transmit power and vice versa. auxiliary information and core frame. and Error Detection) Figure A1.2 depicts the generic structure of the AMR frame. AMR frame structure and operating modes Figure A1. B and C classes. The Frame Type can indicate the use of one of the eight AMR codec modes for that frame. In case of a comfort noise frame. The auxiliary information part includes the Mode Indication. The Frame Quality Indicator indicates if the frame is good or bad. Mode Request and Codec CRC fields.2.e. Generic structure of the AMR frame . The CRC field is used for the purpose of errordetection calculated over all the Class A bits in the AMR Core frame.1. replace “class A” bits of the core frame while “class B” and “class C” bits are omitted. The Core frame part is used to carry the encoded bits divided into A. a SID frame. As observed in the figure. The “optimum” AMR codec mode is dynamically selected as function of the channel quality Speech quality Good Poor Good Channel quality Bad Figure A1.

AMR operating modes Table A1. B and C.1. This class is protected by the CRC in auxiliary information field. Similarly.75 kbps AMR 5.95 kbps AMR 10.70 kbps” mode is equivalent to the codec used in the PDC Japanese standard. Classes B and C contain bits where increasing error rates gradually reduce the speech quality. “AMR 6.2 kbps (GSM EFR) AMR SID GSM EFR SID TDMA EFR SID PDC EFR SID Future usage No data to transmit/receive 42 49 55 58 61 75 65 81 – – – – – – Class B Class C bits bits 53 54 63 76 87 84 99 103 – – – – – – 0 0 0 0 0 0 40 60 – – – – – – Table A1. the “AMR 7. comfort noise. or other) bits index 0 1 2 3 4 5 6 7 8 9 10 11 12-14 15 AMR 4.15 kbps AMR 5. AMR 7.1 depicts the 8 different modes (source bit rates) AMR can operate. Finally.40 kbps (IS-136 EFR).40 kbps” mode is equivalent to the IS-641 codec used in the USA standard IS-136 (US TDMA). Class A contains the bits that are most sensitive to errors and any kind of errors in these bits typically result in a corrupted speech frame which should not be decoded without applying appropriate error concealment. AMR modes and relationship with AMR frame structure . but the decoding of an erroneous speech frame is usually possible without a strongly perceptible quality degradation. The reason for dividing the speech bits into classes is that they can be subjected to different error protection in the network. For instance.AMR Codec in UMTS 377 Classification of the encoded bits according to their sensitivity to errors AMR encoded bits are divided into three indicative classes according to their importance: A.90 kbps AMR 6. It should be noted that some of these modes are equivalent to the speech codecs currently used in other mobile communication systems. Frame content ClassA Frame type (AMR mode.70 kbps (PDC EFR) AMR 7. the “AMR 12.20 kbps” mode is equal to the ETSI GSM called codec EFR (Enhanced Full Rate Speech [TS 06. AMR 12.2 kbps .60]).

might think that their connection is broken during silence periods. the AMR mode control function is part of the RNC. since this control entity needs information from the air interface to make a decision about valid AMR modes – the AMR mode command is used to change the current AMR mode to the new one. logically controlled by Non-Access Stratum protocols. Without this background noise the participants in a conversation.093] techniques where RF transmission is cut during speech pauses. This solution is not applicable in UTRA due to the soft-handover procedure defined for dedicated traffic channels.2. however. The receiver decodes these parameters and generates locally a “comfort noise”. Note that in GSM networks the control of the codec mode is provided by the BTS. the RNC generates the AMR mode adaptation command based on existing radio conditions in the downlink as reported by the UE . this means that all AMR coded data is going to be transmitted not only via Iub and air interface but also via Iu-interfaces. Within the radio interface.378 UMTS Based on the fact that voice activity in a normal conversation is about 40%. VAD works together with the Discontinuous Transmission (DTX) or Source Controlled Rate (SCR) [TS 26. A1. When the transmission is cut. Mode adaptation can independently be applied in the uplink and the downlink. Location of the AMR speech codec in UMTS networks The AMR speech codec is located in the Transcoder (TC) function defined to be in the UMTS core network and as such. From the transfer point of view. AMR mode adaptation in the downlink As shown in Figure A1. “comfort noise” parameters are sent at a regular rate in AMR frames during discontinuous activity. the rate on the speech connection is either decreased or increased depending on the new valid AMR mode by changing the valid Transport Format (TF) in the corresponding MAC-d entity (see Chapter 7). The only element in the network which can provide this type of information is the UTRAN. Therefore. a different AMR mode can be used in each direction and this can be dynamically changed during a voice conversation. that the AMR mode control that generates the AMR mode command cannot be located in the TC. Dynamic AMR mode adaptation The AMR mode adaptation in UMTS networks means using different AMR coding for the data stream. The SCR technique for AMR in UMTS is mandatory and aims at prolonging the battery life (UE side) and reducing the interference. These frames are known as SID (SIlence Descriptor) frames. At any point in time. and more precisely a part of layer 3 functionality.3. Note. all AMR modes implement a Voice Activity Detection (VAD) algorithm that detects if each 20 ms-frame contains speech or not on the transmitting side.

Control of AMR modes Iu-CS UE AMR speech codec Uu Node B Iub RNC TC AMR speech codec AMR encoded speech data in UL (ongoing call) Request to modify the AMR mode in DL AMR encoded speech data in DL with new AMR mode (ongoing call) Change AMR mode Request to modify the AMR mode in UL Change AMR mode AMR encoded speech data in UL with new AMR mode (ongoing call) Figure A1. – In the second proposed alternative. the RNC decides when to request the encoder in the UE to change the valid AMR mode and a new valid AMR mode is sent to the UE inside the AMR mode command message.3.3). which has been communicated to the UE by RRC from the RNC side. The changed AMR mode is discovered by the RNC from the TFCI bits in the uplink dedicated physical data channel. Overview of AMR codec mode control during an ongoing voice call Uplink AMR mode adaptation Two different alternatives for the AMR mode control in the uplink have been proposed: – Based on the air-interface load. when the level of the maximum transmission power is reached. the UE does not have any rights to request the mode adaptation from the network nor to change the used AMR mode autonomously. When received by the UE. be selected only from the valid Transport Format Set (TFS). This enables the UE to change the valid AMR mode of the speech connection on uplink more quickly without requesting the mode change from the RNC first. The new mode can. mode adaptation is made accordingly (see Figure A1. For instance. The command is sent to the encoder inside the TC via the Iu interface. however. Within this approach.AMR Codec in UMTS 379 from radio quality measurements and from traffic volume measurements. . the AMR codec control is not only included into the RNC but also into the UE. the UE may change the valid AMR mode independently.

Resource allocation for an AMR speech connection An AMR speech connection can be initiated either by the UE or the network.2 kbps mode in the radio interface A1.4). Depending on whether the requested AMR base speech connection supports the concept of Unequal Error Protection (UEP) or Equal Error Protection (EEP). When the UE requests resources from the network. The codec is based on the same adaptive . RRC may allocate one or none signaling radio bearer according to the alternative method used to change the AMR mode. respectively. coding 1/3 CRC = 12 bits 303 MAC 103 bits DCH (Class B bits) TTI = 20 ms Conv.4. coding 1/2 CRC = 0 bits 136 128 bits (RM) Physical layer 294 bits (RM) 324 bits (RM) DPCH Data 1 6 TPC TFCI 0 2 Data 2 28 Pilots 4 DPCH slot structure (SF = 128) Figure A1.4. RNC can define the requested RAB and associated Radio Bearer(s) (RB). coding 1/3 CRC = 0 bits 333 60 bits DCH (Class C bits) TTI = 20 ms Conv.3. needed which will be then indicated to the UTRAN inside the RANAP RAB ASSIGNMENT REQUEST message. In the control plane. for the user plane (see Figure A1. Based on this request. NAS radio bearer 1 81 bits radio bearer 2 103 bits RLC (TM) 103 bits DTCH radio bearer 3 60 bits RLC (TM) 60 bits DTCH RLC RLC (TM) 81 bits DTCH 81 bits DCH (Class A bits) TTI = 20 ms Conv. AMR wideband The AMR wideband (AMR-WB) codec has been standardized in 3GPP and is part of the specification in Release 5. The CN will determine the QoS. a first negotiation is made based on NAS procedures in order to configure the call connection. Implementation of AMR 12. the RNC assigns either one or three RBs (including one or three DCHs).380 UMTS A1.

. 14.05 kbps and 23.190. R5]. 23.85 kbps.000 Hz.85 kbps. 19.65 kbps. The AMR-WB comprises nine codec modes: 6.AMR Codec in UMTS 381 principles as the AMR narrowband.85 kbps. The encoder of the AMR-WB is able to code an audio signal with bandwidth between 50 and 7.000 samples/s. A higher sampling rate is thus needed compared with the narrowband approach (16 kHz instead of 8 kHz) leading to a 14 bit samples with 16. Wideband coding provides improved voice quality especially in terms of increased voice naturalness since it covers twice the audio bandwidth compared to the classical telephone voice bandwidth of 4 kHz [TS 26.6 kbps. 8.25 kbps. 15. 12. 18.25 kbps.85 kbps.

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