Advanced audio coding (AAC), part of ISO/MPEG-2, issued as an international standard in April, 1997. It supports single or multiple channel audio programs and delivers excellent audio quality at or below 64 kbps/chann...
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Advanced audio coding (AAC), part of ISO/MPEG-2, issued as an international standard in April, 1997. It supports single or multiple channel audio programs and delivers excellent audio quality at or below 64 kbps/channel by exploiting the compression capabilities of a high-resolution filterbank, backward-adaptive prediction, joint channel coding, nonlinear quantizers and noiseless (Huffman) coding. This paper describes the flexible Huffman coding algorithm used in AAC and discusses the compression provided by this component of the standard.
In this paper a generalized perceptual filter which aims to reduce the audible quantization noise in low bitrate audio coding is presented. The perceptual filter is based on a human auditory perception model whereby i...
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In this paper a generalized perceptual filter which aims to reduce the audible quantization noise in low bitrate audio coding is presented. The perceptual filter is based on a human auditory perception model whereby its derivation takes the highly nonlinear nature of the human ears into account and thus attempts to model the psychoacoustic behaviour of the ear. The results show the perceptual filter can reduce the audible quantization noise in a subband coded audio signal under a subjective analysis. The proposed filter has straightforward implementation, thus making it suitable for incorporation into future coder designs.
Stereophonic coding has been applied to the ISO/MPEG audio coding algorithm to improve the quality of stereo signals at very low bit rates. While reasonable quality has been achieved at bit rates of 2*64 kbit/s or low...
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ISBN:
(纸本)0780308034
Stereophonic coding has been applied to the ISO/MPEG audio coding algorithm to improve the quality of stereo signals at very low bit rates. While reasonable quality has been achieved at bit rates of 2*64 kbit/s or lower, it was found that the proposed scheme could not achieve consistent good quality sounds for all types of audio signals. An improved stereophonic coding scheme is presented. This improved scheme employs the use of error minimisation and power equalization to obtain the improvements. Simulations of the scheme at a bit rate of 128 kbit per stereophonic channel produced a significant improvement in the quality of the audio over the original scheme. A quality that is close to transparency can be obtained at a bit rate of 128 kbit/s.< >
A new bit assignment algorithm is presented. Its goals are the simultaneous assignment on all subbands in a few steps of an iterative calculus, the use of memory to achieve a better speed of convergence and the consid...
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A new bit assignment algorithm is presented. Its goals are the simultaneous assignment on all subbands in a few steps of an iterative calculus, the use of memory to achieve a better speed of convergence and the consideration of a deformable error curve. The basis of the algorithm is discussed and also other considerations that are likely to arise in practice. Finally, an example of its performance is given.
audio coding at low bitrates suffers from artifacts due to spectrum truncation. Typical audio codecs code multi-channel sources using transforms across the channels to remove redundancy such as middle (mid) - side (M/...
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ISBN:
(纸本)9781424423538
audio coding at low bitrates suffers from artifacts due to spectrum truncation. Typical audio codecs code multi-channel sources using transforms across the channels to remove redundancy such as middle (mid) - side (M/S) coding. At low bitrates, the spectrum of the coded channels is truncated and the spectrum of the channels with lower energy, such as the side channel, is truncated severely, sometimes entirely. This results in a muffled sound due to truncation of all coded channels beyond a certain frequency. It also results in a loss of spatial image even at low frequencies due to severe truncation of the side channel. Previously we have developed a low bitrate coding method to combat the loss of higher frequencies caused by spectrum truncation. In this paper, we present a novel low bitrate audio coding scheme to mitigate the loss of spatial image. Listening tests show that the combination of the two low bitrate coding methods results in a audio codec that can get good quality even at bitrates as low as 32 kbps for stereo content with low decoder complexity.
Sinusoidal coding plays an important part in low rate audio coding. Typically, differential techniques are employed to reduce the bit rate for the representation of sinusoidal parameters. We compare several schemes fo...
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Sinusoidal coding plays an important part in low rate audio coding. Typically, differential techniques are employed to reduce the bit rate for the representation of sinusoidal parameters. We compare several schemes for time-differential (TD) and frequency-differential (FD) representation of the sinusoidal model parameters. We show, through simulation experiments, that bit rates obtained with FD techniques are competitive with those achieved using TD techniques, provided that a variable length segmentation algorithm is first applied to the input signal. This result is important because FD techniques (in contrast to TD techniques) do not rely on the presence of previous segments for correct decoding, and therefore offer robust performance in a lossy packet channel environment.
We study the behavior of hybrid random waveform models for audio signals, involving sparse random series of waveforms, with random coefficients. Similar approaches have been considered in the recent years. However, th...
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We study the behavior of hybrid random waveform models for audio signals, involving sparse random series of waveforms, with random coefficients. Similar approaches have been considered in the recent years. However, these do generally not rely on explicit models, and are of more "algorithmical" nature. The models we propose allow us to analyze mathematical properties of such signals and corresponding estimators, and derive estimation algorithms, which do not rely on complex optimization techniques
Perceptual coding techniques have recently been applied successfully to high-quality coding of digital audio signals. The basic perceptual coding system uses an analysis/synthesis system to map the time-domain data in...
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Perceptual coding techniques have recently been applied successfully to high-quality coding of digital audio signals. The basic perceptual coding system uses an analysis/synthesis system to map the time-domain data into a number of frequency-domain channels. A perceptual model is used to estimate the amount of noise shaping needed in order to avoid any audible noise due to the quantization of the frequency domain data. Different filterbanks have been applied to perceptual coding. A comparison of different filterbanks shows that there is no performance penalty for hybrid filterbanks compared to other solutions.< >
This paper proposes a new quantization for transform coefficients based on algebraic quantization. The coefficients are represented by a few pulses multiplied by a unique amplitude. The coefficients to be transmitted ...
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This paper proposes a new quantization for transform coefficients based on algebraic quantization. The coefficients are represented by a few pulses multiplied by a unique amplitude. The coefficients to be transmitted are selected by optimizing an error criterion, that determines the signs, positions and amplitudes of the pulses. This simple quantization has been implemented in a wavelet-based wideband scalable coder, and has been proved to provide a perceptually better quality than SPIHT on speech signal and music.
In this paper, we present a comparative study of different lossless audio coding schemes, which are implemented using different integer transforms. The audio signal under consideration, which is assumed to be integer-...
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ISBN:
(纸本)1424402719;1424402727
In this paper, we present a comparative study of different lossless audio coding schemes, which are implemented using different integer transforms. The audio signal under consideration, which is assumed to be integer-valued, as in the case of fixed-point implementation, is first decorrelated using the appropriate integer transform. The resulting integer coefficients are then entropy-coded to produce the output stream. Several integer transforms have been considered, such as the integer wavelet transform (IntWT) with different decomposition levels as well as different filters, the integer discrete cosine transform (IntDCT), and the integer Walsh Hadamard transform (IntWHT). Arithmetic and Huffman coding have been used for entropy coding. The performed simulation provides insight on the performance of the different integer transforms in the lossless audio coding context
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