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Name: Yashas Prakash Student ID :1000803680

EE 5359 Multimedia Processing Project Proposal Study and implementation of G.719 audio codec and performance analysis of G.719 with AAC (advanced audio codec) and HE-AAC (high efficiency-advanced audio codec) audio codecs . Name: Yashas Prakash Student ID :1000803680

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Name: Yashas Prakash Student ID :1000803680

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  1. EE 5359 Multimedia Processing Project ProposalStudy and implementation of G.719 audio codec and performance analysis of G.719 with AAC (advanced audio codec) and HE-AAC (high efficiency-advanced audio codec) audio codecs. Name: YashasPrakash Student ID :1000803680 Instructor: Dr. K. R. Rao Date: 03-27-2012

  2. Introduction To Codecs • A codec is a device or computer program capable of encoding or decoding a digital data stream or signal. • It can be thought of as a Compressor/De-compressor OR Encode/Decode • Codec programs are required for the media player to play audio/video files. • A codec encodes a data stream or signal for transmission, storage or encryption, or decodes it for playback or editing. • Codecs are used in videoconferencing, streaming media and video editing applications.

  3. Introduction to G.719 Codecs • G.719 is an ITU-T standard audio codec providing high quality, moderate bit rate (32 to 128 kbit/s) wideband (20 Hz - 20 kHz audio bandwidth, 48 kHz audio sample rate) audio coding at low computational load [1]. • It was produced through a collaboration between Polycom and Ericsson. • G.719 incorporates elements of Polycom's Siren22 codec (22 kHz) and Ericsson codec technology, as well as Polycom's Siren7 and Siren14 codecs (G.722.1 and G.722.1 Annex C), which have been used in videoconferencing systems for many years [1].

  4. Advantage and Application of G.719 • The algorithm is designed to provide 20 Hz - 20 kHz audio bandwidth using a 48kHz sample rate, operating at 32 - 128 kbps [3]. • This codec features very high audio quality and low computational complexity and is suitable for use in applications such as videoconferencing, teleconferencing, and streaming audio over the Internet [3].

  5. OVERVIEW OF THE G.719 CODEC • The G.719 codec is a low-complexity transform-based audio codec and can provide an audio bandwidth of 20 Hz to 20 kHz at 32 - 128 kbps. • The codec features very high audio quality and extremely low computational complexity compared to other state-of-the-art audio coding algorithms. • G.719 is optimized for both speech and music. • It is based on transform coding with adaptive time-resolution, adaptive bit-allocation and low complexity lattice vector quantization [1]. • The computational complexity is quite low (18 floating-point MIPS) for an efficient high-quality compressor [1].

  6. G.719 Contd.. • The codec operates on 20 ms frames, and the algorithmic delay end-to-end is 40 ms [2]. • The encoder input and decoder output are sampled at 48 kHz [2].

  7. Block diagram of G.719 encoder Block diagram of the G.719 encoder [1].

  8. ADAPTIVE TIME-FREQUENCY-TRANSFORM • The adaptive time-frequency transform is based on the detection of a transient sounds [3]. • In the case of transient sounds, the time-frequency transform will increase its time resolution and allows a better representation of the rapid changes in the input signal characteristics [3].

  9. G.719 Decoder Block diagram of the G.719 decoder [1].

  10. COMPLEXITY AND MEMORY REQUIREMENTS Table 1: Complexity and memory requirements [4]

  11. Complexity in G.719 • Complexity is a paramount parameter for a codec. Complex codecs require more powerful and more expensive digital signal processors (DSPs) to run on [1]. • This increases the product cost and power consumption, which limits the codec usability [1]. • The fixed-point C-code implementation of G.719, which is an integral part of the Recommendation, is based on a set of instructions that mimics a generic DSP instruction set [1].

  12. An overview of AAC codec • Advanced audio coding(AAC) scheme was a joint development by Dolby, Fraunhoffer, AT&T, Sony and Nokia [9]. • It is a digital audio compression scheme for medium to high bit rates which is not backward compatible with motion pictures experts group (MPEG) audio standards [9]. • AAC is a second generation coding scheme which is used for stereo and multichannel signals. When compared to the perceptual coders, AAC provides more flexibility and uses more coding tools [12].

  13. AAC codec contd., • The AAC encoding follows a modular approach and the standard define four profiles which can be chosen based on factors like complexity of bitstream to be encoded, desired performance and output[9]. • Low complexity (LC) • Main profile (MAIN) • Sample-rate scalable (SRS) • Long term prediction (LTP)

  14. An overview of the HE-AAC codec • High efficiency advanced audio codec is a lossy data compression scheme. • It is an extension of low complexity AAC optimized for low bitrate operations such as streaming audio. • HEAAC using spectral band replication (SBR) technology to enhance the compression efficiency in frequency domain. • Scientific testing by the European Broadcasting Union has indicated that HE-AAC at 48 kbit/s was ranked as "Excellent" quality using the MUSHRA scale. • Testing indicates that material decoded from 64 kbit/s HE-AAC does not yet have similar audio quality to material decoded from MP3 at 128 kbit/s using high quality encoders.

  15. Block diagram of the SBR encoder

  16. Block diagram of the SBR decoder

  17. References • [1] M. Xie, P. Chu, A. Taleb and M. Briand, " A new low-complexity full band (20kHz) audio coding standard for high-quality conversational applications ", IEEE, Workshop on Applications of Signal Processing to Audio and Acoustics, pp.265-268, Oct. 2009. • [2] A. Taleb and S. Karapetkov, " The first ITU-T standard for high-quality conversational fullband audio coding ", IEEE, communications magazine, vol.47, pp.124-130, Oct. 2009. • [3] J. Wang, B. Chen, H. He, S. Zhao and J. Kuang, " An adaptive window switching method for ITU-T G.719 transient coding in TDA domain", IEEE, International Conference on Wireless, Mobile and Multimedia Networks, pp.298-301, Jan. 2011. • [4] J. Wang, N. ning, X. ji and J. kuang, " Norm adjustment with segmental weighted SMR for ITU-T G.719 audio codec ", IEEE, International Conference on Multimedia and Signal Processing, vol.2, pp.282-285, May. 2011.

  18. References • [5] K. Brandenburg and M. Bosi, “ Overview of MPEG audio: current and future standards for low-bit-rate audio coding ” JAES, vol.45, pp.4-21, Jan/Feb. 1997. • [6] A/52 B ATSC Digital Audio Compression Standard: http://www.atsc.org/cms/standards/a_52b.pdf • [7] F. Henn , R. Böhm and S. Meltzer, “ Spectral band replication technology and its application in broadcasting ”, International broadcasting convention, 2003. • [8] M. Dietz and S. Meltzer, “CT-AACPlus – a state of the art audio coding scheme”, Coding Tecnologies, EBU Technical review, July. 2002. • [9] ISO/IEC IS 13818-7, “ Information technology – Generic coding of moving pictures and associated audio information Part 7: advanced audio coding (AAC) ”, 1997.

  19. References • [10] M. Bosi and R. E. Goldberg, “ Introduction to digital audio coding standards ”, Norwell, MA, Kluwer, 2003. • [11] H. S. Malvar, “ Signal processing with lapped transforms ”, Artech House, Norwood MA, 1992. • [12] D. Meares, K. Watanabe and E. Scheirer, “ Report on the MPEG-2 AAC stereo verification tests ”, ISO/IEC JTC1/SC29/WG11, Feb. 1998.

  20. THANK YOU

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