14th European Signal Processing Conference (EUSIPCO 2006), Florence, Italy, September 4-8, 2006, copyright by EURASIP

Similar documents
3GPP TS V6.2.0 ( )

MPEG-4 aacplus - Audio coding for today s digital media world

Parametric Coding of Spatial Audio

ETSI TS V (201

3GPP TS V ( )

Audio-coding standards

ELL 788 Computational Perception & Cognition July November 2015

ISO/IEC INTERNATIONAL STANDARD. Information technology MPEG audio technologies Part 3: Unified speech and audio coding

The MPEG-4 General Audio Coder

Audio-coding standards

5: Music Compression. Music Coding. Mark Handley

ETSI TS V8.0.0 ( ) Technical Specification

Optical Storage Technology. MPEG Data Compression

Parametric Coding of High-Quality Audio

INTERNATIONAL ORGANISATION FOR STANDARDISATION ORGANISATION INTERNATIONALE DE NORMALISATION ISO/IEC JTC1/SC29/WG11 CODING OF MOVING PICTURES AND AUDIO

Mpeg 1 layer 3 (mp3) general overview

MPEG-4 General Audio Coding

Audio coding for digital broadcasting

The following bit rates are recommended for broadcast contribution employing the most commonly used audio coding schemes:

SAOC and USAC. Spatial Audio Object Coding / Unified Speech and Audio Coding. Lecture Audio Coding WS 2013/14. Dr.-Ing.

ETSI TS V ( )

Audio Compression. Audio Compression. Absolute Threshold. CD quality audio:

25*$1,6$7,21,17(51$7,21$/('(1250$/,6$7,21,62,(&-7&6&:* &2',1*2)029,1*3,&785(6$1'$8',2 &DOOIRU3URSRVDOVIRU1HZ7RROVIRU$XGLR&RGLQJ

Lecture 16 Perceptual Audio Coding

MPEG-4 Structured Audio Systems

INTERNATIONAL ORGANISATION FOR STANDARDISATION ORGANISATION INTERNATIONALE DE NORMALISATION ISO/IEC JTC1/SC29/WG11 CODING OF MOVING PICTURES AND AUDIO

Efficiënte audiocompressie gebaseerd op de perceptieve codering van ruimtelijk geluid

Modeling of an MPEG Audio Layer-3 Encoder in Ptolemy

Module 9 AUDIO CODING. Version 2 ECE IIT, Kharagpur

New Results in Low Bit Rate Speech Coding and Bandwidth Extension

DAB. Digital Audio Broadcasting

The Reference Model Architecture for MPEG Spatial Audio Coding

Technical PapER. between speech and audio coding. Fraunhofer Institute for Integrated Circuits IIS

MPEG-1. Overview of MPEG-1 1 Standard. Introduction to perceptual and entropy codings

Convention Paper Presented at the 121st Convention 2006 October 5 8 San Francisco, CA, USA

Scalable Perceptual and Lossless Audio Coding based on MPEG-4 AAC

Efficient Representation of Sound Images: Recent Developments in Parametric Coding of Spatial Audio

6MPEG-4 audio coding tools

Development of Low Power ISDB-T One-Segment Decoder by Mobile Multi-Media Engine SoC (S1G)

MPEG-4 BSAC Technology

Chapter 14 MPEG Audio Compression

ETSI TS V ( )

Appendix 4. Audio coding algorithms

EE482: Digital Signal Processing Applications

ISO/IEC Information technology High efficiency coding and media delivery in heterogeneous environments. Part 3: 3D audio

Perceptual Coding. Lossless vs. lossy compression Perceptual models Selecting info to eliminate Quantization and entropy encoding

_äìé`çêé. Audio Compression Codec Specifications and Requirements. Application Note. Issue 2

Principles of Audio Coding

Optimized architectures of CABAC codec for IA-32-, DSP- and FPGAbased

Fast frame memory access method for H.264/AVC

ENTROPY CODING OF QUANTIZED SPECTRAL COMPONENTS IN FDLP AUDIO CODEC

I D I A P R E S E A R C H R E P O R T. October submitted for publication

The BroadVoice Speech Coding Algorithm. Juin-Hwey (Raymond) Chen, Ph.D. Senior Technical Director Broadcom Corporation March 22, 2010

Multimedia Communications. Audio coding

Multi-Pulse Based Code Excited Linear Predictive Speech Coder with Fine Granularity Scalability for Tonal Language

Compressed Audio Demystified by Hendrik Gideonse and Connor Smith. All Rights Reserved.

Audio Engineering Society. Convention Paper. Presented at the 126th Convention 2009 May 7 10 Munich, Germany

Audio Fundamentals, Compression Techniques & Standards. Hamid R. Rabiee Mostafa Salehi, Fatemeh Dabiran, Hoda Ayatollahi Spring 2011

Upcoming Video Standards. Madhukar Budagavi, Ph.D. DSPS R&D Center, Dallas Texas Instruments Inc.

Perceptual coding. A psychoacoustic model is used to identify those signals that are influenced by both these effects.

WHITE PAPER. Fraunhofer Institute for Integrated Circuits IIS

ATRAC DSP Type-R. Sound compression Technique for Mini Disc

An Ultra Low-Power WOLA Filterbank Implementation in Deep Submicron Technology

Parametric Audio Based Decoder and Music Synthesizer for Mobile Applications

Principles of MPEG audio compression

Enhanced Audio Features for High- Definition Broadcasts and Discs. Roland Vlaicu Dolby Laboratories, Inc.

Proceedings of Meetings on Acoustics

Perceptual Audio Coders What to listen for: Artifacts of Parametric Coding

Both LPC and CELP are used primarily for telephony applications and hence the compression of a speech signal.

Perceptually motivated Sub-band Decomposition for FDLP Audio Coding

ISO/IEC INTERNATIONAL STANDARD

Real Time Implementation of TETRA Speech Codec on TMS320C54x

AUDIOVISUAL COMMUNICATION

Audio Coding and MP3

MPEG-4 ALS International Standard for Lossless Audio Coding

Perceptual Pre-weighting and Post-inverse weighting for Speech Coding

MAXIMIZING AUDIOVISUAL QUALITY AT LOW BITRATES

Video Compression An Introduction

Lecture 5: Error Resilience & Scalability

Convention Paper 8654 Presented at the 132nd Convention 2012 April Budapest, Hungary

Presents 2006 IMTC Forum ITU-T T Workshop

FINE-GRAIN SCALABLE AUDIO CODING BASED ON ENVELOPE RESTORATION AND THE SPIHT ALGORITHM

Implementation of G.729E Speech Coding Algorithm based on TMS320VC5416 YANG Xiaojin 1, a, PAN Jinjin 2,b

Efficient Implementation of Transform Based Audio Coders using SIMD Paradigm and Multifunction Computations

Design and Implementation of 3-D DWT for Video Processing Applications

Audio Processing on ARM Cortex -M4 for Automotive Applications

DVB Audio. Leon van de Kerkhof (Philips Consumer Electronics)

IO [io] MAYAH. IO [io] Audio Video Codec Systems

SAOC FOR GAMING THE UPCOMING MPEG STANDARD ON PARAMETRIC OBJECT BASED AUDIO CODING

Preview only reaffirmed Fifth Avenue, New York, New York, 10176, US

MPEG-4 Version 2 Audio Workshop: HILN - Parametric Audio Coding

SPEEX CODEC IMPLEMENTATION ON THE RPB RX210

MPEG-4 Advanced Audio Coding

MPEG-1 Bitstreams Processing for Audio Content Analysis

DRA AUDIO CODING STANDARD

AudioLiquid Converter User Guide

Efficient Signal Adaptive Perceptual Audio Coding

Spatial Audio Object Coding (SAOC) The Upcoming MPEG Standard on Parametric Object Based Audio Coding

3GPP TS V7.1.0 ( )

Specification for the use of Video and Audio Coding in DVB services delivered directly over IP protocols

Transcription:

TRADEOFF BETWEEN COMPLEXITY AND MEMORY SIZE IN THE 3GPP ENHANCED PLUS DECODER: SPEED-CONSCIOUS AND MEMORY- CONSCIOUS DECODERS ON A 16-BIT FIXED-POINT DSP Osamu Shimada, Toshiyuki Nomura, Akihiko Sugiyama and Masahiro Serizawa Media and Information Research Laboratories, NEC Corporation 1753 Shimonumabe, Nakahara-ku, Kawasaki, 216-8666, Japan phone: + (81) 44 431 7513, fax: + (81) 44 431 7590, email: o-shimada@bx.jp.nec.com web: www.nec.com Abstract This paper investigates tradeoff between complexity and memory size in the 3GPP enhanced aacplus decoder based on 16-bit fixed-point DSP implementation. In order to investigate this tradeoff, the speed- and the-memory conscious decoders are implemented. The maximum number of operations for the implemented speed-conscious decoder is million cycles per second (MC) for a 32 kb/s bitstream. The maximum number of operations for the memoryconscious decoder, where 70% of the data are allocated to an external memory area, increases by 5.7 MC (19%) for the bitstream. The investigation of this tradeoff provides an actual relationship between the computational complexity and the internal memory size of the 3GPP enhanced aacplus decoder. The implemented decoders enable music download and streaming services on next-generation mobile terminals. 1. INTRODUCTION The 3rd generation partnerships project (3GPP) [1] has already standardized many technical specifications for 3G mobile systems. In these specifications, highly efficient speech and audio codecs have been included. 3GPP enhanced aacplus [2] is a newly standardized audio codec for mobile applications. It enables high quality stereo audio compression at a low bitrate in the range of 24 32 kb/s thanks to integration of MPEG-4 parametric stereo () [3][4] into MPEG-4 high efficiency advanced audio coding (HE-) [5]. When 3GPP enhanced aacplus decoder is implemented on a DSP for music download and streaming services on mobile terminals, there is a tradeoff between low computational complexity and low internal memory size. However, this tradeoff in the 3GPP enhanced aacplus decoder has not been reported. This paper investigates tradeoff between complexity and memory size in the 3GPP enhanced aacplus decoder based on 16-bit fixed-point DSP implementation. In Section 2, the 3GPP enhanced aacplus algorithm is explained in reference to the HE- algorithm. Section 3 demonstrates computational complexity of the implemented decoder with and without an external memory area on the DSP in order to investigate this tradeoff. 3GPP enhanced aacplus MPEG-4 MPEG-4 HE- Additional decoding tools MPEG-4 > Error concealment > Stereo-to-mono downmix MPEG-4 > Spline resampler Fig. 1. Framework of 3GPP enhanced aacplus 2. 3GPP enhanced aacplus Figure 1 shows a framework of 3GPP enhanced aacplus. It is a standard audio codec integrating MPEG-4 into HE-. enables audio coding at a low bitrate for stereo signals by efficiently representing stereo information. Sound quality of 3GPP enhanced aacplus at 24 kb/s is statistically equivalent to that of HE- at 32 kb/s [6]. Additional decoding tools are specified in 3GPP surrounded by dashed lines in Fig. 1. They are error concealment tools, stereo-tomono downmix tools and a spline resampler tool. 2.1 MPEG-4 HE- standard MPEG-4 HE- is a standard extending MPEG-4 advanced audio coding () [7] by incorporating spectral band replication () [5][8]. Figure 2 depicts an encoder and a decoder block diagram of HE- shown in Fig. 1. An encoder and a decoder are a pre- and a postprocessor combined with an encoder and a decoder, respectively. At the encoder in Fig. 2 (a), the encoder splits an input signal into low- and high-frequency components. The low-frequency components are used to synthesize a band-limited signal, which is encoded as low-frequency information at the encoder. The high-frequency components are used to calculate a spectral envelope in the encoder. The envelope and other parameters are encoded at a low bitrate as high frequency information. The low- and the high-frequency information are multiplexed into the output bitstream.

Band-limited Demultiplexer High-freq. Low-freq. (a) Low-freq. High-freq. (b) Multiplexer Band-limited Fig. 2. Block diagrams of the HE- encoder and decoder. Mono signal Demultiplexer Stereo Information HE- Mono (a) Mono HE- Stereo Information (b) Multiplexer Mono Fig. 3. Block diagrams of the 3GPP enhanced aacplus encoder and decoder. At the decoder shown in Fig. 2 (b), the bitstream demultiplexer separates the input bitstream into the low- and the high-frequency information. The decoder generates the band-limited signal from the low-frequency information. The high-frequency information is used to expand the bandwidth of the band-limited signal in order to improve sound quality at the decoder. The expansion process operates in two modes, namely, high quality (HQ- ) and low power (LP-). LP- has been developed specially for mobile terminals to reduce battery consumption. It employs real-valued operations instead of complex-valued operations used in HQ-. However, the real-valued operations cause several problems that seriously deteriorate sound quality. For improving sound quality in real-valued operations, LP- incorporates additional modules [9] with low computational complexity. As a result, The HE- decoder with LP- achieves 30 % lower computational complexity than that with HQ- without statistical difference in sound quality [10]. 2.2 3GPP enhanced aacplus standard Figure 3 shows block diagrams of the 3GPP enhanced aacplus encoder and decoder. At the encoder in Fig. 3 (a), the encoder separates the stereo signal into stereo information and a mono signal that is obtained by down-mixing the stereo signal. The stereo information consists mainly of the inter-channel intensity difference (IID) and the interchannel coherence (ICC). IID and ICC represent the power difference and the correlation between the left and the right signals, respectively. The mono signal is encoded as mono information by the HE- encoder. The stereo information and the mono information are multiplexed into the output bitstream. At the decoder in Fig. 3 (b), the input bitstream is demultiplexed into the mono and the stereo information. The HE- decoder generates the mono signal by decoding the mono information. The decoded mono signal is extended to the stereo signal using the stereo information in the decoder. Demultiplexer Error Concealment Stereo-tomono Downmix Spline Resampler Analysis Filter Bank High Freq. Generator Envelope Adjuster Synthesis Filter Bank Fig. 4. Block diagram of the 3GPP enhanced aacplus decoder. In the specification of 3GPP enhanced aacplus1, two modes are standardized for stereo signals. At a bitrate higher than 36 kb/s, the encoder is the same as the HE- encoder illustrated in Fig. 2 (a). In this case, the decoder operates in the LP- mode to reduce computational complexity. On the other hand, the framework illustrated in Fig.3 (a) is used to encode stereo signals at lower than or equal to 36 kb/s for improving sound quality. 2.3 3GPP enhanced aacplus decoder Figure 4 shows a detailed block diagram of the 3GPP enhanced aacplus decoder. The input bitstream is demultiplexed into the data, the data and the data. The decoder generates the mono signal from the data. An analysis filter bank splits the mono signal into mul- 1 Both the encoder and the decoder are standardized different from the MPEG standard that specifies only the decoder.

tiple low frequency subband signals. High frequency subband signals are generated in a high frequency generator by copying the low-frequency subband signals. Energy of the generated high frequency subband signals is adjusted in an envelope adjuster using the data. The low and high frequency subband signals in the mono channel are provided for the decoder. It generates decorrelated subband signals that have no correlation with the mono subband signals. By mixing the mono and the decorrelated subband signals with a ratio indicated by IID and ICC, stereo subband signals are obtained. A synthesis filter bank synthesizes the stereo signal from these stereo subband signals. The decoder utilizes complex-valued operations for reducing aliasing problems caused by modifying lowfrequency subband signals. This is because the human auditory system is sensitive to distortions in the low frequency. The decoder used with the decoder also utilizes complex-valued operations, since the and the decoders are connected with the complex-valued subband signals. Three modules surrounded by dashed lines in Fig. 4 are the additional modules specified in 3GPP. The error concealment tools improve sound quality in case of a frame loss. The error concealment tool requires the status of the previous frame and that of the next frame. Consequently, the delay of the decoder increases by one frame. The and the error concealment tools generate parameters from those in the previous frames. The stereo-to-mono downmix tools are used for monaural terminals. The data are mixed from stereo to mono in the parameter domain. In this case, the decoder works as the mono decoder in order to reduce computational complexity. The spline resampler tool decimates the output signal for terminals with a narrowband D-to-A converter. This tool converts the sampling frequency from 22.05 or 24 khz to 8 or 16 khz. 3. DSP IMPLEMENTAION For the mobile terminals, it is essential to implement its decoder on an LSI with low power consumption. This is equivalent to DSP implementation with a least number of operations per second. Furthermore, an internal memory area of the DSP is usually shared with other applications such as a video decoder and an acoustic post processing. Therefore, it is of great importance to minimize the internal memory size of the decoder. The balance of the speed and the memory should be determined by mobile terminal manufacturers or their applications. In view of this tradeoff, two decoders have been implemented on a 16-bit fixed-point DSP, TMS320VC5510 [11], running at 200 MHz, namely, speed-conscious and memoryconscious implementations. The speed-conscious implementation uses only the internal memory area of the DSP to minimize the number of operations by eliminating the access latency to the external memory area. The memoryconscious implementation minimizes the internal memory size of the decoder at the expense of a slight increase in the number of operations. Both decoders were developed by Number of operations [MC] 40 30 20 10 0 stereo HE- (LP) 19.3 27.2 13.3 13.9 mono HE- (HQ) +29% +85% 13.5 15.8 48 kb/s 32 kb/s 48 kb/s 32 kb/s Average Maximum Fig. 5. Computational complexity of the speed-conscious decoder. Tab. 1. Memory size of the speed-conscious decoder [kword] Static RAM 19.0 Scratch RAM 13.5 Constant ROM 19.0 Program ROM 37.4 Total 88.9 incorporating and additional tools into the HE- decoder described in [6]. The decoding process has been optimized by maximizing the parallel use of ALUs and MACs, in order to achieve low computational complexity. 3.1 Performance evaluation of speed-conscious decoder The number of operations was evaluated with two bitstreams encoded at 48 kb/s and 32 kb/s. They were generated, by the 3GPP floating-point reference encoder [12], from a typical stereo music signal sampled at 48 khz. Complexity of these bitstreams was measured by the 3GPP fixedpoint reference decoder [13]. The maximum numbers of operations were 32.2 and 34.6 wmo 2 (weighed million operations per second) for the 48 kb/s and the 32 kb/s bitstreams, respectively. Figure 5 shows the computational complexity of the implemented speed-conscious decoder with no frame losses. Based on the enhanced aacplus specification, the 48 kb/s bitsream was decoded by stereo HE- with LP-. was used for decoding the 32 kb/s bitstream in combination with mono HE- with HQ-. In this evaluation, the stereo-to-mono downmix and the spline resampler tools in Fig. 4 were disabled. All the data and the instructions code are allocated in the internal memory area on the DSP. 2 wmo is a measure of computational complexity used in ETSI and ITU-T. This measure is roughly equivalent to MC (million cycles per second) for fixed-point DSPs.

48 kb/s 32 kb/s Speed-conscious decoder Memory-conscious decoder 28.7 35.0 +6.0 (26%) +5.7 (19%) Tab. 2. Memory allocation of the memory-conscious decoder [kword] Internal External Total Static RAM 4.8 14.2 19.0 Scratch RAM 12.8 0.7 13.5 Constant ROM 8.0 11.0 19.0 Program ROM 0 37.4 37.4 Overall 25.6 63.3 88.9 0 10 20 30 Number of operations [MC] Fig. 6. Complexity comparison between the memory- and the speed-conscious decoders. The average numbers of operations for the 48 and the 32 kb/s bitstreams are 19.3 and 27.2 MC, respectively. 27.2 MC consists of 13.9 MC for mono HE- with HQ- and 13.3 MC for. The complexity of is about the same as that of mono HE- with HQ-. The maximum numbers of operations are 23.1 and 33.7 MC for the 48 and the 32 kb/s bitstreams, respectively. 33.7 MC consists of 16.2 MC for mono HE- with HQ- and 17.5 MC for. Consequently, the 3GPP enhanced aacplus decoder can be implemented with 85% increase in computational complexity compared to the mono HE- decoder with HQ-. It is 29% more complex than the stereo HE- decoder with LP-. The memory size of the implemented speed-conscious decoder is shown in Tab. 1. The sizes of static RAM, scratch RAM, constant ROM and program ROM are 19.0, 13.5, 19.0 and 37.4 kword, respectively. Overall, the memory size is 88.9 kword. Scratch RAM is shared with other applications, since data in the area are used to decode only the current frame. The size of constant ROM is larger than that in 3GPP reference fixed-point decoder [10]. This is because several data in constant ROM such as huffman codebooks have been modified to achieve low computational complexity. The codebooks are also divided into sub-codebooks to extract quantized values efficiently. Program ROM keeps instruction code to decode bitstreams. 3.2 Performance evaluation of memory-conscious decoder In order to reduce the internal memory use as well as the increase in computational complexity, some data were moved to the external memory area. These data are not accessed frequently in the decoding process. All the instruction data in program ROM are allocated to the external memory area, since the complexity increase can be avoided by use of the instruction cache. In addition to the moved data mentioned earlier, some areas in static RAM such as filter buffers were moved to the external memory area. Although these data are accessed frequently, the access is limited to a specific period. To reduce the latency in the memory access, the data are copied to the internal memory area just before the RAM access. After the access, they are stored to the external memory area again. Table 2 presents the memory allocation of the memoryconscious decoder. The internal memory sizes of static RAM, scratch RAM, constant ROM and program ROM are 4.8, 12.8, 8.0 and 0 kword, respectively. Since 70 % of the data was moved to the external memory area, the internal memory size is reduced from 88.9 kword to 25.6 kword. The remaining internal memory area and computing power can be used for other applications. The computational complexity of the memory-conscious decoder was evaluated in comparison with that of the speedconscious decoder. The conditions are the same as the earlier evaluation. Figure 6 illustrates the computational complexity of the memory- and the speed-conscious decoders. The maximum numbers of operations for the 48 kb/s and the 32 kb/s bitstreams are 28.7 and 35.0 MC, respectively. 35.0 MC consists of 19.8 MC for mono HE- with HQ- and 15.2 MC for. The maximum number of operations for the 48 kb/s and the 32 kb/s bitstreams increase by 6.0 and 5.7 MC, respectively. While 70% of the data are allocated to the external memory area, the maximum numbers of operations increase only by 26% and 19% for the 48 kb/s and the 32 kb/s streams, respectively. 3.3 Tradeoff between complexity and internal memory size Figure 7 summarizes the tradeoff between the computational complexity and the internal memory size in the 3GPP enhanced aacplus decoder for the 32 kb/s bitstream. The area surrounded by dashed oval, which is distributed from top left to bottom right, shows the feasible area of the decoder. The position in the area is determined based on the decoder implementation. The speed-conscious decoder is located at the bottom right, since it is optimized for the computational complexity. The position of the memory-conscious decoder is not the top left, because complexity drastically increases when the internal memory size is further reduced. The balance of the speed and memory size should be determined by the application. For cellular phones, the memory-conscious decoder is preferable, since the decoder is usually used with other applications. On the other hand, the speed-conscious

Complexity [MC] 35.0 Memoryconscious Speedconscious 25.6 88.9 Internal memory size [kword] Fig. 7. Tradeoff between complexity and internal memory size for the 32 kb/s bitstream. 96 kb/s 67% HE- 48 kb/s 33% 3GPP enhanced aacplus 32 kb/s 20.0 +6.6 (30%) +9.3 (47%) 0 10 20 30 Number of operations [MC] Fig. 8. Complexity comparison among the, the HE- and the 3GPP enhanced aacplus decoders. decoder is suitable for music-only players in order to reduce battery consumption. Both of the implemented decoders enable music download and streaming services on nextgeneration mobile terminals. 3.4 Complexity comparison Computational complexity of the implemented 3GPP enhanced aacplus decoder was compared with that of conventional and HE- decoders. The, the HE- and the 3GPP enhanced aacplus bitstreams were encoded at 96, 48 and 32 kb/s from typical stereo signals sampled at 48 khz. These bitrates were chosen so that the bandwidth of each decoded stereo signal is about 16 khz. Figure 8 shows the comparison result on computational complexity among three decoders. All the data and the instructions code of each decoder are allocated in the internal memory area on the DSP. The maximum numbers of operations are 20.0, and MC for the, the HE- and the 3GPP enhanced aacplus decoders, respectively. 3GPP enhanced aacplus achieves 67% bitrate reduction from at the cost of 47% increase in the number of operations. Furthermore, 3GPP enhanced aacplus achieves 33% bitrate reduction from HE- at the cost of 30% increase in computational complexity. 4. CONCLUSIONS Tradeoff between complexity and memory size in the 3GPP enhanced aacplus decoder based on 16-bit fixed-point DSP implementation has been investigated. It has been shown that the maximum number of operations for the implemented speed-conscious decoder is and MC for the 48 and 32 kb/s bitstreams. It has also been shown that the maximum number of operations for the memoryconscious decoder, where 70% of the data are allocated to the external memory area, increases only by 6.0 and 5.7 MC for the bitstreams, respectively. The investigation of this tradeoff demonstrates the relationship between the computational complexity and the internal memory size of the 3GPP enhanced aacplus decoder. The implemented decoders enable music download and streaming services on nextgeneration mobile terminals. REFERENCES [1] http://www.3gpp.org/ [2] 3GPP TS 26.401, General audio codec audio processing General description. [3] ISO/IEC 14496-3:2001/Amd 2:2004, Parametric coding for high-quality audio. [4] Erik Schuijers, et al., Low complexity parametric stereo coding, 6073, the 116th AES Convention. [5] ISO/IEC 14496-3:2001/Amd 1:2003, Bandwidth extension. [6] ISO/IEC JTC 1/SC 29 /WG 11, N7137, Listening test report on MPEG-4 High Efficiency v2. [7] ISO/IEC 14496-3:2001, Information technology - Coding of audio-visual objects Part 3: Audio. [8] Martin Dietz, et al., Spectral band replication, a novel approach in audio coding, 5553, the 112th AES Convention. [9] Osamu Shimada, et al., A Low Power for the MPEG-4 Audio Standards and its DSP implementation, 6048, the 116th AES Convention. [10] ISO/IEC JTC1/SC29/WG11, N6009, Report on the Verification Tests of MPEG-4 High Efficiency. [11] Texas Instruments, SPRS076L, TMS320VC 5510/5510A Fixed-Point Digital Processors. [12] 3GPP TS 26.410, General audio codec audio processing Floating-point ANSI-C code. [13] 3GPP TS 26.411, General audio codec audio processing Fixed-point ANSI-C code.