Dsp Based Audio Compression Research Papers (original) (raw)
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Recent papers in Dsp Based Audio Compression
MPEG stands for MOVING PICTURE EXPERTS GROUP is a standard for video and audio compression for eliminating the noisy signals from the transmitted signals from the satellite. Audio compression is a basic method defined under MPEG-1 and... more
MPEG stands for MOVING PICTURE EXPERTS GROUP is a standard for video and audio compression for eliminating the noisy signals from the transmitted signals from the satellite. Audio compression is a basic method defined under MPEG-1 and MPEG-4 which by coding techniques compress audio signals to filter out undesired signals.This paper focuses on the MPEG technology, need and coding technique for audio compression.
We outline a method to perform efficient low rate quantization for MPEG-4 advanced audio coding (AAC). The AAC bit stream consists of indices for quantized spectral coefficients as well as side information about quantizer step sizes and... more
We outline a method to perform efficient low rate quantization for MPEG-4 advanced audio coding (AAC). The AAC bit stream consists of indices for quantized spectral coefficients as well as side information about quantizer step sizes and Huffman codebooks. The MPEG-4 Verification Model does not explicitly account for side information bits in its optimization and suffers from poor compression efficiency at low bit rates. We reformulate the encoding problem as one of optimal parameter selection, where the side information bits are taken into account, so as to minimize the noise to mask ratio for the given target bit rate. The optimal solution is determined by a dynamic programming procedure that efficiently searches through a trellis. This trellis-based optimization greatly improves the low bit rate performance of AAC and, consequently, the performance of a multi-layer AAC system. The resulting bit stream is standard-compatible, and additional complexity due to the proposed optimization is only incurred at the encoder
In this paper, we present results for a combined wavelet warped linear prediction (WLP) audio coder. In contrast to conventional LP, WLP allows for the control of frequency resolution to closely match the response of the human auditory... more
In this paper, we present results for a combined wavelet warped linear prediction (WLP) audio coder. In contrast to conventional LP, WLP allows for the control of frequency resolution to closely match the response of the human auditory system. The coder first uses WLP analysis on each frame of audio, and then applies a discrete wavelet transform (DWT) to the residual signal (prediction error). A psychoacoustic model is used in parallel to obtain a global masking threshold used in bit allocation. Bits are dynamically allocated to the DWT coefficients in an attempt to minimise the perceptually significant quantisation error. For monophonic signals sampled at 44.1 kHz, the coder achieves near transparent quality for a variety of speech and music signals at an average bit-rate of 64 kb/s. The power of the proposed coder resides in its easy scalability to lower bit rates