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UT Arlington EE 5359 - AC-3, AAC and HE-AAC audio codecs

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EE5359 MULTIMEDIA PROCESSING REPORTProject Proposal:Title: Study and comparison of AC-3, AAC and HE-AAC audio codecsAbstract:The spectral band replication technology (SBR) is an advancement in the field of low bit rateaudio coding and it enhances the performance of the traditional audio coders. CodingTechnologies, an international company in the audio coding field has developed and marketedSBR. MPEG-AAC belonging to the ISO-MPEG standard has shown a tremendous improvementwith SBR.[1] The coding efficiency of the traditional audio coders with SBR increases at least by30%.[7] The SBR is a bandwidth extension technique which exploits the strong correlation effectbetween the low and high frequency content in an audio signal. In this project, a performanceanalysis of the MPEG-AAC audio coders and advanced audio coding (AAC) audio coders withSBR will be implemented which includes a comparison of the coding efficiency. Student: Dhatchaini Rajendran Student ID: 1000636681Email: [email protected] Date: September 28, 2010ACRONYMS AND ABBREVIATIONSAAC - Advanced audio codingAC-3 – Audio codec 3AES - Audio Engineering SocietyATSC – Advanced television systems committeeHE-AAC – High efficiency advanced audio codingIMDCT – Inverse modified discrete cosine transformISO – International organization for standardizationLC – Low complexityLFE – Low frequencies enhancementLTP – Long term predictionMDCT – Modified discrete cosine transformMPEG – Moving pictures experts groupPCM – Pulse code modulationSBR – Spectral band replicationSRS – Sample rate scalableTNS – Temporal noise shapingAn Overview of Perceptual Audio CodingAudio coding algorithms aim at representing the audio signal with minimum number of bits andat the same time achieves signal reproduction with minimum errors. Perceptual audio coding algorithms make use of facts like the insensitivity of the human ear tofrequencies less than 20 kHz and the redundancy in audio signals to accomplish maximumcompression of the audio signal. The irrelevant information in the signal is identified by usingseveral psychoacoustic parameters like absolute hearing thresholds, simultaneous masking,critical band frequency analysis, temporal masking and spread of masking along the basilarmembrane. Digital Audio InputFigure 1: Block diagram of perceptual encoding/ decoding scheme [1]The blocks in Fig.1 are explained below:- The filter bank decomposes the digital input signal into its subsampled spectralcomponents in the time or frequency domain. - The perceptual model uses the time domain input signal and mostly the output of theanalysis filter bank along with the psychoacoustic rules, and calculates the actualmasking threshold. This is called the perceptual model of the perceptual encoding system.- The quantization and coding of the spectral components is done and the noise introduced by quantizing below the masking threshold level is retained. There are several ways of accomplishing this step from simple block companding to analysis-by-synthesis systems using additional noiseless compression.- A bitstream formatter is used in the encoding of the bitstream which is made up ofquantized and coded spectral coefficients and some side information like bit allocationinformation.Analysis Filter BankPerceptual ModelQuantization and CodingEncoding of BitstreamAn Overview of AC-3 Audio CodecAC-3 is an audio codec developed by Dolby Laboratories. Dolby AC-3 audio compressionalgorithm is a advanced television systems committee (ATSC) standard for digital audiocompression.[2] It is a lossy audio compression format and supports multi-channel format and isused in a variety of applications including digital television and DVD.There are 5 full range channels (3Hz- 20,000Hz). Three of them are in the front (left, right andcentre) and the other two are surround channels. The sixth channel ranges from 3Hz-120Hz andis also known as low frequencies enhancement (LFE) Channel. This set of channels is known as“5.1” channels. Figure 2: Block diagram of AC-3 encoder [2]The working of the AC-3 encoder blocks in Fig. 2 is explained here [2]. Transforming the representation of audio from a sequence of PCM time samples into a sequence of frequency coefficients blocks is the first step in the encoding process. This is accomplished with the analysis filter bank. Overlapping blocks of 512 time samples are transformed into the frequency domain by multiplying them with a time window. As the blocks overlap, each PCM input sampleis represented by two sequential transformed blocks. Thus the frequency domain representation gets decimated by a factor of two and so each block will contain 256 frequency coefficients. A binary exponent and mantissa is used to represent each frequency. The set of exponents isencoded into a coarse representation of the signal spectrum which is referred to as the spectral envelope. The core bit allocation routine is used to determine the number of bits used to encode each individual mantissa. The mantissa is then quantized according to the bit allocation information. The spectral envelope and the coarsely quantized mantissas for 6 audio blocks (1536 audio samples) are formatted into an AC-3 frame. The AC-3 bit stream (from 32 to 640 kbps) is a sequence of AC-3 frames. The AC-3 decoder function is the exact opposite to the encoder.An overview of MPEG – Advanced Audio CodingAdvanced audio coding scheme was a joint development by Dolby, Fraunhoffer, AT&T, Sonyand Nokia.[9] It is a digital audio compression scheme for medium to high bit rates which is notbackward compatible with motion pictures experts group (MPEG) audio standards. The AACencoding follows a modular approach and the standard define four profiles which can be chosenbased on factors like complexity of bitstream to be encoded, desired performance and output. - Low complexity (LC)- Main profile (MAIN) - Sample-rate scalable (SRS)- Long term prediction (LTP)Excellent audio quality is provided by AAC and it is suitable for low bit rate high quality audioapplications. MPEG – AAC audio coder uses the AAC scheme.HE – AAC also known as aacPlus is a low bit rate audio coder. It is an AAC LC audio coderenhanced with SBR technology. A generic block diagram of an AAC encoder is


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UT Arlington EE 5359 - AC-3, AAC and HE-AAC audio codecs

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