Lucent Technologies Inc.

At&t Bell Laboratories

Other

Alcatel Lucent

Juin-Hwey Chen

Craig R Watkins

Joseph Gerard Aguilar

Wei Wang

Robert W Zopf

Richard Vandervoort Cox

Nuggehally Sampath Jayant

David Anthony Campana

Robert J McAulay

Xiaoqin Sun

Robert B Dunn

Xiaoquin Sun

Vipul Parikh

A system and method for processing of audio and speech signals is disclosed, which provide compatibility over a range of communication devices operating at different sampling frequencies and/or bit rates. The analyzer of the system divides the input signal in different portions, at least one of which carries information sufficient to provide intelligible reconstruction of the input signal. The analyzer also encodes separate information about other portions of the signal in an embedded manner, so that a smooth transition can be achieved from low bit-rate to high bit-rate applications. Accordingly, communication devices operating at different sampling rates and/or bit-rates can extract corresponding information from the output bit stream of the analyzer. In the present invention embedded information generally relates to separate parameters of the input signal, or to additional resolution in the transmission of original signal parameters. Non-linear techniques for enhancing the overall performance of the system are also disclosed. Also disclosed is a novel method of improving the quantization of signal parameters. In a specific embodiment the input signal is processed in two or more modes dependent on the state of the signal in a frame. When the signal is determined to be in a transition state, the encoder provides phase information about N sinusoids, which the decoder end uses to improve the quality of the output signal at low bit rates.

Scalable and embedded codec for speech and audio signals

A system and method are provided for processing audio and speech signals using a pitch and voicing dependent spectral estimation algorithm (voicing algorithm) to accurately represent voiced speech, unvoiced speech, and mixed speech in the presence of background noise, and background noise with a single model. The present invention also modifies the synthesis model based on an estimate of the current input signal to improve the perceptual quality of the speech and background noise under a variety of input conditions. The present invention also improves the voicing dependent spectral estimation algorithm robustness by introducing the use of a Multi-Layer Neural Network in the estimation process. The voicing dependent spectral estimation algorithm provides an accurate and robust estimate of the voicing probability under a variety of background noise conditions. This is essential to providing high quality intelligible speech in the presence of background noise.

Parametric speech codec for representing synthetic speech in the presence of background noise

A system and method are provided for processing audio and speech signals using a pitch and voicing dependent spectral estimation algorithm (voicing algorithm) to accurately represent voiced speech, unvoiced speech, and mixed speech in the presence of background noise, and background noise with a single model. The present invention also modifies the synthesis model based on an estimate of the current input signal to improve the perceptual quality of the speech and background noise under a variety of input conditions. The present invention also improves the voicing dependent spectral estimation algorithm robustness by introducing the use of a Multi-Layer Neural Network in the estimation process. The voicing dependent spectral estimation algorithm provides an accurate and robust estimate of the voicing probability under a variety of background noise conditions. This is essential to providing high quality intelligible speech in the presence of background noise. In one embodiment, the waveform coding is implemented by separating the input signal into at least two sub-band signals and encoding one of the at least two sub-band signals using a first encoding algorithm to produce at least one encoded output signal; and encoding another of said at least two sub-band signals using a second encoding algorithm to produce at least one other encoded output signal, where the first encoding algorithm is different from the second encoding algorithm. In accordance with the described embodiment, the present invention provides an encoder that codes N user defined sub-band signal in the baseband with one of a plurality of waveform coding algorithms, and encodes N user defined sub-band signals with one of a plurality of parametric coding algorithms. That is, the selected waveform/parametric encoding algorithm may be different in each sub-band.

Method and system for sub-band hybrid coding

High-quality, low-complexity and low-delay scalable and embedded system and method are disclosed for coding speech and general audio signals. The invention is particularly suitable in Internet Protocol (IP)-based multimedia communications. Adaptive transform coding, such as a Modified Discrete Cosine Transform, is used, with multiple small-size transforms in a given signal frame to reduce the coding delay and computational complexity. In a preferred embodiment, for a chosen sampling rate of the input signal, one or more output sampling rates may be decoded with varying degrees of complexity. Multiple sampling rates and bit rates are supported due to the scalable and embedded coding approach underlying the present invention. Further, a novel adaptive frame loss concealment approach is used to reduce the distortion caused by packet loss in communications using IP networks.

Low-complexity, low-delay, scalable and embedded speech and audio coding with adaptive frame loss concealment

A tunable perceptual weighting filter is used in tandem codecs (coder/decoders). Specific filter parameters are advantageously tuned to provide improved performance in tandeming contexts. The parameters used are 10th order LPC (Linear Predictive Coding) predictor coefficients. The system employed uses Low-Delay Code Excited Linear Predictive codecs (LD-CELP).

Tunable perceptual weighting filter for tandem coders

A speech compression system called 'Transform Predictive Coding', or TPC, provides for encoding 7 kHz wideband speech (16 kHz sampling) at a target bit-rate range of 16 to 32 kb/s (1 to 2 bits/sample). The system uses short-term and long-term prediction to remove the redundancy in speech. A prediction residual is transformed and coded in the frequency domain to take advantage of knowledge in human auditory perception. The TPC coder uses only open-loop quantization and therefore has a fairly low complexity. The speech quality of TPC is essentially transparent at 32 kb/s, very good at 24 kb/s, and acceptable at 16 kb/s.

Synthesis of speech signals in the absence of coded parameters

A speech coding system robust to frame erasure (or packet loss) is described. Illustrative embodiments are directed to a modified version of CCITT standard G.728. In the event of frame erasure, vectors of an excitation signal are synthesized based on previously stored excitation signal vectors generated during non-erased frames. Specifically, the decoder generates and stores samples of a first excitation signal in a memory, and then, in response to a signal indicating a frame erasure, the decoder synthesizes a second excitation signal based on the previously stored samples. In particular, the second excitation is synthesized by correlating a first subset of the stored samples with a second subset thereof, identifying a set of stored excitation signal samples based on the correlation, and synthesizing the second excitation signal based on the identified samples. Finally, the decoder then filters the second excitation signal to synthesize a signal reflecting human speech.

Linear prediction coefficient generation during frame erasure or packet

A speech compression system called 'Transform Predictive Coding', or TPC, provides for encoding 7 kHz wideband speech (16 kHz sampling) at a target bit-rate range of 16 to 32 kb/s (1 to 2 bits/sample). The system uses short-term and long-term prediction to remove the redundancy in speech. A prediction residual is transformed and coded in the frequency domain to take advantage of knowledge in human auditory perception. The TPC coder uses only open-loop quantization and therefore has a fairly low complexety.

Perceptual noise masking measure based on synthesis filter frequency

A highly efficient, low delay pitch parameter derivation and quantization permits overall delay which is a fraction of prior coding delays for equivalent speech quality at low bitrates. In distinguishing between pitch period information for voiced and non-voiced frames of input signals, non-voiced frames are assigned a non-zero 'bias' value, while voiced frames have associated with them generated pitch information based on an analysis of signals in a present frame and comparison with signals relating to the pitch in a prior frame. Transitions from non-voiced to voiced input frames are efficiently accomplished using a non-uniform quantization method based on an analysis of a sequence of frames. Typical uses include low delay, low-bitrate coders such as Code Excited Linear Prediction (CELP).

Pitch period estimation for use with audio coders

A speech coding system robust to frame erasure (or packet loss) is described. Illustrative embodiments are directed to a modified version of CCITT standard G.728. In the event of frame erasure, vectors of an excitation signal are synthesized based on previously stored excitation signal vectors generated during non-erased frames. This synthesis differs for voiced and non-voiced speech. During erased frames, linear prediction filter coefficients are synthesized as a weighted extrapolation of a set of linear prediction filter coefficients determined during non-erased frames. The weighting factor is a number less than 1. This weighting accomplishes a bandwidth-expansion of peaks in the frequency response of a linear predictive filter. Computational complexity during erased frames is reduced through the elimination of certain computations needed during non-erased frames only. This reduction in computational complexity offsets additional computation required for excitation signal synthesis and linear prediction filter coefficient generation during erased frames.

Computational complexity reduction during frame erasure of packet loss

A speech compression system called 'Transform Predictive Coding', or TPC, provides for encoding 7 kHz wideband speech (160 kHz sampling) at a target bit-rate range of 16 to 32 kb/s (1 to 2 bits/sample). The system uses short-term and long-term prediction to remove the redundancy in speech. A prediction residual is transformed and coded in the frequency domain to take advantage of knowledge in human auditory perception. The TPC coder uses only open-loop quantization and therefore has a fairly low complexity. The speech quality of TPC is essentially transparent at 32 kb/s, very good at 24 kb/s, and acceptable at 16 kb/s.

Speech signal quantization using human auditory models in predictive

An adaptive postfilter is used on the decoding side of tandem codecs (coder/decoders). Post-filter parameters are adapted using a backward synthesis filter. The parameters used are 10th order LPC (Linear Predictive Coding) predictor coefficients. The system employed uses Low-Delay Code Excited Linear Predictive codecs (LD-CELP).

Tunable post-filter for tandem coders

Codebook vectors may be considered critical if they give poor energy approximations and exhibit a particular shape with smaller components near the beginning and larger components toward the end of the vector. Standard deviation may be used to identify critical codevectors based on energy approximation error measured in decibels. A low-bit rate (typically 8 kbit/s or less), low-delay digital coder and decoder based on Code Excited Linear Prediction for speech and similar signals features backward adaptive adjustment for codebook gain and short-term synthesis filter parameters and forward adaptive adjustment of long-term (pitch) synthesis filter parameters. In addition, the coder makes use of an excitation codebook and the coding is based on a set of codebook vector energies for a set of codebook vectors in the codebook. The codebook energies are calculated by identifying a set of approximations for the non-critical codebook vector energies. This achieves a significant reduction in processing time in comparison with prior art techniques.

Energy calculations for critical and non-critical codebook vectors

A low-bitrate (typically 8 kbit/s or less), low-delay digital coder and decoder based on Code Excited Linear Prediction for speech and similar signals features backward adaptive adjustment for codebook gain and short-term synthesis filter parameters and forward adaptive adjustment of long-term (pitch) synthesis filter parameters. A highly efficient, low delay pitch parameter derivation and quantization permits overall delay which is a fraction of prior coding delays for equivalent speech quality at low bitrates.

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Neshanic Station, NJ, United States of America

Juin-Hwey Chen