Design and performance of robust embedded ADPCM coder

An embedded adaptive DPCM (EADPCM) speech coder is described which allows bit rate reductions to be achieved by progressive deletion of bits from output codewords. Optimised step size multipliers are given for a robust implementation using an improved algorithm for adaptive quantisation. Simulation shows that a graceful reduction in speech quality with bit rate is achieved in the range 16-48 kbit/s.<>     

Embedded DPCM for Variable Bit Rate Transmission

Within the bit stream of an embedded digital code is a stream that can be decoded to produce a reasonable replica of the analog source signal. Unlike pulse code modulation (PCM), differential PCM (DPCM), is not an embedded code. IfCbits/sample are delected from the bit stream of a DPCM encoder withEbits/sample, the decoded analog signal is substantially noisier than the output of a DPCM codec withD = E - Cbits/sample. The penalty is 4-10 dB in signal-to-noise ratio (snr). However, with minor modifications to the encoder and decoder, DPCM becomes an embedded code. Embedded DPCM withEbits/ sample at the encoder andDbits/sample transmitted produces exactly the same output as embedded DPCM withDbits/sample encoding and perfect transmission. The snr of embedded DPCM is slightly lower than the Snr of DPCM. The penalty is 0.5-0.8 dB if the minimum transmitted bit rate is 2 bits/sample. It is less than 0.3 dB ifDis at least 3 bits/sample. Combined with an appropriate adaptive quantizer the embedded DPCM codec produces embedded ADPCM (adaptive DPCM) for variable rate transmission ranging from 2 bits/sample up to any desired maximum. Applications exist in speech interpolation, packet switching, and hardware architecture.     

Determination of optimum reconstruction levels for embeddeddecoders

An iterative technique is described for computing optimum decoder reconstruction levels from input statistics given from I to R received bits per sample of the output of an R-bit per sample linear PCM encoder. Simulation of a DPCM embedded coding scheme with speech input shows that significant SNR gains are possible by using optimum reconstruction levels for decoding the supplementary bit stream     

Highly Sensitive Speech Detector and High-Speed Voiceband Data Discriminator in DSI-ADPCM Systems

This paper describes a highly sensitive speech detector and a high-speed voiceband data classifier capable of discriminating between speech and voiceband data of a 4.8 kbit/s 8-phase PSK and 4.8 kbit/s 8-point QAM, and a 9.6 kbit/s 16-point QAM as described in a CCITT recommendation. The presence of a speech signal is detected by analyzing short-time energies, zero-crossing rates, and sign bit sequences of the input signal. The proposed speech detector, with a short hangover time of 32 ms, is able to reduce the average talk spurt activity in an international satellite link to 36 percent. This detector can also classify the detected speech into narrow-band or wide-band spectrum sounds or a low power sound for a variable rate ADPCM encoding. Discrimination between speech and high-speed voiceband data is based on short-time energies, a zeros-crossing rate and linear prediction coefficients of an adaptive predictor. Classification among a 4.8 kbit/s 8-phase PSK and 8-point QAM, and a 9.6 kbit/s 16-point QAM can also be performed by an average prediction gain and a coefficient of variation of the short-term amplitude distribution of the input signal. Discrimination of voiceband data was performed successfully, and erroneous discrimination of talk spurt of telephone speech as voiceband data were, respectively, four times for two two-party conversations lasting 5 minutes in an international satellite link. This is equivalent to less than 0.09 percent of the conversation time.     

Transparent Wideband Data Transmission in Digital Communication Networks

This paper presents a systematic study of the possibilities of time-transparent and of code-transparent binary data transmission in digital communication networks where PCM or differential PCM (DPCM) systems are employed for the encoding of speech or video signals. To describe the effects that occur if data signals instead of speech or video signals are sampled and transmitted by these two encoding methods, time-transparency and code-transparency are defined and are used as figures of merit. The occurrence of time-quantizing errors and amplitude errors at the decoder output is statistically analyzed. The limitations to be considered in the various applications are described, and time- and code-transparency are evaluated quantitatively. The predictions of the analysis are then compared with the results obtained through simulations. The comparison shows that for data rates below 10 percent of the bit rate (bits per second) in the digital channel, conventional PCM and DPCM encoding are acceptable for binary data transmission and have the advantage of easy signal integration. For higher data rates special data encoding is necessary. The encoding method analyzed here is dual-mode.     

Entropy coding of ITU G.728 speech parameters for low bit rateaudiovisual communications

Entropy coding principles are applied to the 16 kbit/s ITU G.728 speech codec. It is shown that the average bit rate can be reduced to 14.5 kbit/s without a significant increase in the codec complexity. In very low bit rate audiovisual communication applications such as the videophone, the saved bits can be used to improve the output video quality     

Low delay coder (< 25 ms) of wideband audio (20 Hz-15 kHz) scalable from 64 to 32 kbit/s

A low delay coder for speech and music signals sampled at 32kHz is described. Its algorithmic delay does not exceed 25 ms which enables audioconferencing applications without echo cancellation. Its bit rate is scalable between 64 and 32 kbit/s by steps of 8 kbit/s. The transmitter issues the binary code at 64 kbit/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 music coders. The perceptual signal is obtained in the time domain, then transformed to the 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 with amplitude correction. At the phase 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.     

High-Gain Digital Speech Interpolation with Adaptive Differential PCM Encoding

This paper describes a digital speech interpolationadaptive differential PCM bit reduction technique in which digital speech interpolation (DSI) is combined with ADPCM encoding. A highly sensitive speech detector, a voiceband data discriminator, and a variable rate ADPCM encoding are used to achieve a high compression ratio. The speech detector proposed in [1] detects speech signals above -51 dBm with 32 ms hangover time; average talk spurt activity of 36 percent was measured on fully loaded trunks in an international satellite link. Features of the speech power spectrum are used for adaptively controlling the bit length from 2 to 4 in an ADPCM speech encoder. Voiceband data are detected with 10 ms by the voiceband data discriminator. 5 bit ADPCM encoding is applied to voiceband data to maintain transparency through the DSI-ADPCM system. A DSI gain of 3 is expected as a result of the highly sensitive speech detection, the variable rate encoding technique, and the voiceband data discrimination. Speech and voiceband data are efficiently transmitted through an ADPCM encoding with either a 6 or 6.4 kHz sampling rate converted from an 8 kHz sampling rate. To avoid a band limitation as much as possible, a frequency shift manipulation on the voiceband channel is incorporated prior to the sampling conversion. Consequently, a total bit reduction gain of 7 to 4 is expected relative to a 64 kbit/s PCM transmission. Satisfactorily high quality of the processed speech has been obtained through computer simulations.     

Waveform encoding of voiceband data signals

This is a tutorial paper on encoding the waveform of voiceband data signals into digital form for digital transmission. The encoding algorithms covered are PCM, DPCM, and DM with and without quantizer adaptation. Coder bit rates from 16 to 64 kb/s are considered. Most digital Codec's on voiceband channels are designed primarily for speech and their performance with speech is well known. However, since many voiceband channels may contain other signals such as voiceband data, adequate performance with such signals is a necessity. Much of the paper is a summary and an interpretation of previous work in encoding data signals. Some new perspectives and material are presented, especially for DPCM. Topics covered include the sampling phase effect, fast and slow acting quantizer adaptation algorithms, comparison between speech and data performance, dual-mode coders, network considerations, and the effect of quantizing noise on the phase integrity of data signals.     

Algorithm of Adaptive Bit Allocation Wavelet Transform Audio Coding

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