Speech and audio coding standards al-E defined in international organizations having wide range of activities, only part of them dealing with bit rate compression of speech and audio. The UIT (international Telecommun...
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Speech and audio coding standards al-E defined in international organizations having wide range of activities, only part of them dealing with bit rate compression of speech and audio. The UIT (international Telecommunication Union) deals with interactive communication standards, the ETSI (European Telecommunications Standards Institute) with mobile communication standards in Europe while multimedia communication standard's are under the responsability of the ISO (International Organization Sor Standardization). After a brief description of the standardization mechanism, we will review the features of speech and audio compression schemes (bit rates, quality complexity and delay) and the main applications of these standards. A list of these compression standards, already;adopted or in the course of definition, will be provided for each normalization organization. Finally: this payer gives the orientations or trends which are emerging in the field of audio compression standardization.
The MPEG-2 Advanced Audio Coder is the latest issue of the MPEG audio encoders/decoders family whose most popular version is known as MP3. It gathers many of the latest highly efficient sound compression techniques in...
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The MPEG-2 Advanced Audio Coder is the latest issue of the MPEG audio encoders/decoders family whose most popular version is known as MP3. It gathers many of the latest highly efficient sound compression techniques in a quite classically structured coder. The main part is based or a Discrete Cosine Transform with variable resolution. The output from this filterbank is compressed by the combination of an adaptive bit allocation module, according to frequency subbands, and a set of noiseless Huffman codebooks. Bit allocation is controlled by a psychoacoustic model which determines an audibility threshold far signal distortion in the frequency domain. This article intends to explain the ISO standard without replacing it, and also to be a general introduction to perceptual audio coding.
A low delay coder for speech and music signals sampled at 32 kHz is described. Its algorithmic delay does not exceed 25 ms which enables audio conferencing applications without echo cancellation. Its bot rate is scala...
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A low delay coder for speech and music signals sampled at 32 kHz is described. Its algorithmic delay does not exceed 25 ms which enables audio conferencing applications without echo cancellation. Its bot rate is scalable between 64 and 32 kbits/s by steps of 8 kbits/s. The transmitter issues the binary code at 64 kbits/s with lower bit rate codes embedded in it. The receiver may operate at lower bit rates with gradual loss of quality. The proposed coder is based on a mixed scheme: the adopted solution contains elements from the CELP speech coder and frequency domain where bit allocation is calculated and transform coefficients are quantized. A first solution based on the DFT is discussed, then a second solution based on a MDCT with small overlap is applied. The quantization of these coefficients is done in the following way. First, a prediction of the whole spectrum is applied. Then, a mean-removed gain-shape split VQ is used for amplitude spectrum quantization and a hierarchical 2-dimensional VQ is used for phase spectrum quantization stage, each codeword describing the selected vector index is split into parts corresponding to different bit rates. Due to the hierarchical codebook structure, truncated indices may be used, without much affecting the signal quality. Simulation results are presented and the robustness of the proposed coder is examined.
This paper presents a scalable three bit-rates (8, 14.1 and 24 il kbit/s) coder: For the two embedded lowest ones, operating in the telephone bandwidth, CELP coding techniques are used. For the highest rate, that both...
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This paper presents a scalable three bit-rates (8, 14.1 and 24 il kbit/s) coder: For the two embedded lowest ones, operating in the telephone bandwidth, CELP coding techniques are used. For the highest rate, that both improves narrowband quality and extends the band to [50-7000Hz], transform coding techniques are used. The main applications deal with transmission over network,with no guaranteed QoS.
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