This paper explores the application of the error resilient coding scheme. Absolutely addressed Picture ELement (APEL) coding, to image transmission over noisy radio channels. To improve the performance of APEL at low ...
This paper explores the application of the error resilient coding scheme. Absolutely addressed Picture ELement (APEL) coding, to image transmission over noisy radio channels. To improve the performance of APEL at low signal-to-noise ratios, turbo coding is introduced into the system. Demonstrated through Gaussian and Rayleigh fading channel simulations, this novel combination will be shown to correct and restrict the propagation of the majority of errors incurred during transmission.
In this paper we present a scaleable audio coder, suitable for both speech and music compression. Scalability has been implemented on both bandwidth and bit rate. The encoder accepts 16 and 32 kHz input sampling frequ...
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In this paper we present a scaleable audio coder, suitable for both speech and music compression. Scalability has been implemented on both bandwidth and bit rate. The encoder accepts 16 and 32 kHz input sampling frequencies and operates at 1 bit/sample. In addition to these rates, at bandwidths of 8, 12 and 16 kHz a multi-stage embedded quantization is employed to produce a higher output quality at the expense of an increased bit rate. The design of the algorithm combines subband decomposition and a pulse excitation search offering low complexity, low algorithmic delay and good output quality when compared to existent coding standards.
The proceedings contains 17 papers from the ieecolloquium on Audio and Music Technology: The Challenge of Creative DSP. Topics discussed include: wideband speech and audio coding;audio compression using wavelets;low-...
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The proceedings contains 17 papers from the ieecolloquium on Audio and Music Technology: The Challenge of Creative DSP. Topics discussed include: wideband speech and audio coding;audio compression using wavelets;low-bit-rate frequency extension coding;K ring compression coding;digital sound synthesis;three dimensional stereo sound systems;acoustic echo cancellation;multiway audio distribution over the Internet;sound spatialization;binaural audio processors;three-dimensional sound positioning;real-time singing synthesis;polyphonic music transcription;musical instrument character map synthesis;sound representation;and circular membranes synthesis.
In this paper, we design a simple and efficient VLSI architecture for a novel very fast high performance three step search (FHTSS) algorithm that is superior to the existing three step search (TSS) algorithm in all ca...
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In this paper, we design a simple and efficient VLSI architecture for a novel very fast high performance three step search (FHTSS) algorithm that is superior to the existing three step search (TSS) algorithm in all cases and also to the recently proposed new three step search (NTSS) algorithm when used for lowbit-rate video coding, as with the H.261 standard. Based on a VLSI tree processor and an FPGA addressing circuit, the proposed architecture can successfully implement the FHTSS algorithm on silicon with the minimum number of gates. Because of the flexibility of the architecture, it can also be extended to implement other three step search algorithms.
There are now a number of applications, most notably streamed Internet audio, which require audio and speech to be encoded at a lowbit rate, typically 16 kbit/sec or below. To achieve an acceptable quality, the origi...
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There are now a number of applications, most notably streamed Internet audio, which require audio and speech to be encoded at a lowbit rate, typically 16 kbit/sec or below. To achieve an acceptable quality, the original signal is normally low-pass filtered to somewhere between 4 and 5.5 kHz before encoding. Rather than discard the upper frequency band completely, we propose encoding just the noise component of it using about 500 bits/sec. This greatly enhances contemporary music and close-microphone speech, but has little effect on classical music. The process can be used to enhance any audio or speech codec, knowing only its encoding/decoding delay.
In this paper a new method is shown using the CMV chip-set hardware architecture for the implementation of a high-speed, lowbit-rate imagecoding system. A simple and fast algorithm is introduced to generate basis fu...
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ISBN:
(纸本)0780348672
In this paper a new method is shown using the CMV chip-set hardware architecture for the implementation of a high-speed, lowbit-rate imagecoding system. A simple and fast algorithm is introduced to generate basis functions of 2 dimensional (2D) orthogonal transformations. Using these 2D basis functions of the Hadamard or Cosine functions, the transformation coefficients of the basic blocks of the image are measured by the CNN. Meanwhile, the CNN can produce the inverse transformation of the measured coefficients and the actual distortion-rate can be computed. Ifa required distortion-rate is reached, the coding process could be stopped (the use of even more coefficients would increase bit-rate needlessly). Effects of noise and VLSI computing accuracy are also considered to optimise the architecture. Here we also give a short description how to join the transform coding method and the object-oriented image model.
We present a scaleable audio coder, suitable for both speech and music compression. Scalability has been implemented on both the bandwidth and bit rate. The encoder accepts 16 and 32 kHz input sampling frequencies and...
We present a scaleable audio coder, suitable for both speech and music compression. Scalability has been implemented on both the bandwidth and bit rate. The encoder accepts 16 and 32 kHz input sampling frequencies and operates at 1 bit/sample. In addition to these rates, at bandwidths of 8, 12 and 16 kHz a multi-stage embedded quantisation is employed to produce a higher output quality at the expense of an increased bit rate. The design of the algorithm combines subband decomposition and a pulse excitation search offering low complexity, low algorithmic delay and good output quality when compared to existent coding standards.
There are now a number of applications, most notably streamed Internet audio, which require audio and speech to be encoded at a lowbit rate, typically 16 kbit/sec or below. To achieve an acceptable quality, the origi...
There are now a number of applications, most notably streamed Internet audio, which require audio and speech to be encoded at a lowbit rate, typically 16 kbit/sec or below. To achieve an acceptable quality, the original signal is normally low-pass filtered to somewhere between 4 and 5.5 kHz before encoding. Rather than discard the upper frequency band completely, we propose encoding just the noise component of it using about 500 bits/sec. This greatly enhances contemporary music and close-microphone speech, but has little effect on classical music. The process can be used to enhance any audio or speech codec, knowing only its encoding/decoding delay.
low-delay techniques are proposed for coding 7 kHz speech using subband code-excited linear predictive coding (CELP). The use of separate and joint index codebooks is compared. Specifically, the joint-index-subband CE...
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low-delay techniques are proposed for coding 7 kHz speech using subband code-excited linear predictive coding (CELP). The use of separate and joint index codebooks is compared. Specifically, the joint-index-subband CELP (JISBC) algorithm is found to provide good quality with processing delay in the range 2.375-3.375 ms at corresponding bit rates of 16-8 k bit/s.
A new image compression approach is proposed in which variable block size technique is adopted, using quadtree decomposition, for codingimages at lowbit rates. In the proposed approach, low-activity regions, which u...
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A new image compression approach is proposed in which variable block size technique is adopted, using quadtree decomposition, for codingimages at lowbit rates. In the proposed approach, low-activity regions, which usually occupy large areas in an image, were coded with a larger block size and the block mean is used to represent each pixel in the block, To preserve edge integrity, the classified vector quantisation (CVQ) technique is used to code high-activity regions. A new edge-oriented classifier without employing any thresholds is proposed for edge classification. A novel predictive noiseless coding (NPNC) method which exploits the redundancy between neighbouring blocks is also presented to efficiently code the mean values of low-activity blocks and the addresses of edge blocks. The bit rates required for coding the mean values and addresses can be significantly reduced by the proposed NPNC method. Experimental results show that excellent reconstructed images and higher PSNR were obtained.
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