Speech coding

About: Speech coding is a research topic. Over the lifetime, 14245 publications have been published within this topic receiving 271964 citations.

Data transmitting method, data receiving method, data transmitting device and data receiving device

Abstract: This invention provides a data transmission method and a data receiving method which enable audio data to be multiplexed into video data and be transmitted using a DVI standard cable or the like satisfactorily with a simple configuration. From a data transmitting end, a superimposed video/audio data signal in which audio data are superimposed over a video blanking interval of video data in superimposition timing that is generated using a video blank sync signal and a pixel clock, are transmitted to a data receiving end through the DVI cable, together with the video blank sync signal and the pixel clock. On the data receiving end, a timing signal for extracting the audio data from the superimposed video/audio data signal is generated using the transmitted video blank sync signal and pixel clock, and the superimposed video/audio data signal is separated into video data and audio data using the generated timing signal, as well as the digital audio data are converted into an analog audio signal using an audio clock that is generated by dividing the frequency of the pixel clock.

Method and apparatus employing audio frequency offset extraction and floating-point conversion for digitally encoding and decoding high-fidelity audio signals

Abstract: An audio signal is initially represented by a series of high-resolution pulse code modulated (PCM) data. A lower rate series of representative values are extracted from the initial series of PCM data. Half of the lower rate is at an intermediate audio frequency so that the lower rate series encodes low frequency components of the audio signal. The PCM data are adjusted by offsetting in accordance with corresponding representative values and are then converted to a floating-point representation by extracting scale factor or exponents. The combination of the series of representative values and the floating-point data provides a rate-compressed representation of the audio signal which is capable of being decoded after transmission or storage to reproduce the audio signal without substantial noise, distortion or loss of dynamic range. The splitting of the audio information between the lower rate series and the adjusted floating-point PCM limits the normally destructive effect that low frequency components of high amplitude have upon high frequency components of relatively low amplitude. In a preferred embodiment, a common offset is determined for each block by computing the arithmetic mean of the maximum and minimum PCM data values for the block and truncating the result, the PCM data are adjusted by subtracting their corresponding common offsets, and a common exponent is determined for the block of adjusted PCM data. For encoding high-fidelity audio, preferably the audio signal is initially represented by a series of 16-bit PCM samples at a rate of at least 36 kilohertz, the block size is chosen to be 16 audio samples, and the encoded and compressed data for each block includes a 160 bit frame consisting of an 8-bit block offset, a 3-bit block exponent, a 5-bit error correction code, and sixteen floating-point values each including eight data bits and one parity bit. This format permits 9 stereo audio channels and frame synchronization to be readily transmitted over a conventional video channel.

Audio coding system using characteristics of a decoded signal to adapt synthesized spectral components

Abstract: A receiver in an audio coding system receives a signal conveying frequency subband signals representing an audio signal. The subband signals are examined to assess one or more characteristics of the audio signal. Spectral components are synthesized having the assessed characteristics. The synthesized spectral components are integrated with the subband signals and passed through a synthesis filterbank to generate an output signal. In one implementation, the assessed characteristic is temporal shape and noise-like spectral components are synthesized having the temporal shape of the audio signal.

Audio system parameter setting based upon operator usage patterns

Abstract: A technique for setting parameters of an audio system initially monitors audio related operator usage patterns. The technique then controls an audio source of the audio system based upon the operator usage patterns.

Efficient encoding of speech LSP parameters using the discrete cosine transformation

Abstract: The intraframe and interframe correlation properties are used to develop two efficient encoding algorithms for speech line spectrum pair (LSP) parameters. The first algorithm (2-D DCT), which requires relatively large coding delays, is based on two-dimensional (time and frequency) discrete cosine transform coding techniques; the second algorithm (DCT-DPCM), which does not need any coding delay, uses one-dimensional discrete cosine transform in the frequency domain and DPCM (differential pulse-code modulation) in the time domain. The performances of these systems for different bit rates and delays are studied, and appropriate comparisons are made. It is shown that an average spectral distortion of approximately 1 dB/sup 2/ can be achieved with 21 and 25 bits/frame using the 2-D DCT and DCT-DPCM schemes, respectively. This is a noticeable improvement over the previously reported bit rates of 32 bits/frame and above. >