Koninklijke Philips Corporation N.v.

U.S. Philips Corporation

Andreas Johannes Gerrits

Rakesh Taori

Robert J Sluijter

Albertus Cornelis Den Brinker

Samir Chennoukh

Erik Gosuinus Petrus Schuijers

Arnoldus Werner Johannes Oomen

Gerard Herman Hotho

Marek Zbigniew Szczerba

Arnoldus W J Oomen

Marc Klein Middelink

Dzevdet Burazerovic

Ludovicus M G M Tolhuizen

Arnoldus Werner Johannnes Oomen

Christophe Alain Bernard Hoeppe

Constant Paul Marie Josef Baggen

A device () is arranged for synthesizing sound represented by sets of parameters, each set comprising noise parameters (NP) representing noise components of the sound and optionally also other parameters representing other components, such as transients and sinusoids. Each set of parameters may correspond with a sound channel, such as a MIDI voice. In order to reduce the computational load, the device comprises a selection unit () for selecting a limited number of sets from the total number of sets on the basis of a perceptual relevance value, such as the amplitude or energy. The device further comprises a synthesizing unit () for synthesizing the noise components using the noise parameters of the selected sets only.

Sound synthesis

A decoder receives () a bitstream comprising an encoded mono signal and stereo data. A time scale processor () generates a time scaled mono signal. A time-to frequency processor generates frequency sample blocks of the time scaled signal, the block length being fixed and independent of the time scaling. A parametric stereo decoder () generates a stereo signal for the frequency sample blocks and these are converted to the time domain by a frequency-to-time processor (). A synchronization processor () synchronizes the stereo data with the time scaled signal by determining a time association between a parameter value and a frequency sample block. The parameter value and time association is used to determine synchronized stereo parameter values for that and other frequency sample blocks. The invention is particularly suitable for low complexity generation of time scaled stereo signals from MPEG-4 encoded signals.

Method and apparatus for time scaling of a signal

Coding of an audio signal (x) represented by a respective set of sampled signal values (x(t)) for each of a plurality of sequential time segments is disclosed. The sampled signal values are analyzed to determine one or more sinusoidal components for each of the plurality of sequential segments. The sinusoidal components are linked across a plurality of sequential segments to provide sinusoidal tracks, where each track comprises a number of frames. An encoded signal (AS) is generated, including sinusoidal codes (C) comprising a representation level (r) for each frame or including sinusoidal codes (C) where some of these codes comprise a phase (&#x3c6;), a frequency (&#x3c9;) and a quantization table (Q) for a given frame when the given frame is designated as a random-access frame. The invention allows random access in a track while avoiding long adaptation of the quantization accuracy in a quantizer and/or the need for a large bit stream while still maintaining improved audio quality.

Audio encoding

Coding of an audio signal represented by a respective set of sampled signal values for each of a plurality of sequential segments is disclosed. The sampled signal values are analyzed () to determine one or more sinusoidal components for each of the plurality of sequential segments. The sinusoidal components are linked () across a plurality of sequential segments to provide sinusoidal tracks. For each sinusoidal track, a phase comprising a generally monotonically changing value is determined and an encoded audio stream including sinusoidal codes (r) representing said phase is generated ().

Audio coding via creation of sinusoidal tracks and phase determination

A device for synthesizing sound having sinusoidal components includes a selector for selecting a limited number of the sinusoidal components from each of a number of frequency bands using a perceptual relevance value. The device further includes a synthesizer for synthesizing the selected sinusoidal components only. The frequency bands may be ERB based. The perceptual relevance value may involve the amplitude of the respective sinusoidal component, and/or the envelope of the respective channel.

In a sinusoidal audio encoder a number of sinusoids are estimated per audio segment. A sinusoid is represented y frequency, amplitude and phase. Normally, phase is quantised independent of frequency The invention uses a frequency dependent quantisation of phase, and in particular the low frequencies are quantised using smaller quantisation intervals than at higher frequencies. Thus, the unwrapped phases of the lower frequencies are quantised more accurately, possibly with a smaller quantisation range, than the phases of the higher frequencies. The invention gives a significant improvement in decoded signal quality, especially for low bit-rate quantisers.

Low bit-rate audio encoding

In a sinusoidal audio encoder a number of sinusoids are estimated per audio segment. A sinusoid is represented by frequency, amplitude and phase. The invention uses a track dependent quantization of phase. A track is encoded with a suitable initial (e.g. frequency dependent) quantization grid that is chosen among a set of possible initial grids that may vary from fine to coarse. If, in a series of time segments the frequency variation in a particular track is smaller than a predetermined value, the track is quantized using a finer quantization grid. The invention gives a significant improvement in decoded signal quality, especially for low bit-rate quantizers.

The method creates an audio stream comprising tracks of sinusoidal components linked across a plurality of sequential time segments. Segments in each track are weighted with a normal window (WI, W, W), and consecutive segments have a normal period of overlap () of their trailing edges and leading edges. Segments in which a transientcomponent is determined are weighted with a first modified window (WIm) having a modified trailing edge, and the following segment in the track is weighted with a second modified window (Wm) having a modified leading edge, so that the modified trailing edge and the modified leading edge have a modified period of overlap (m) that comprises the transient component and that is shorter than the normal period of overlap (), and wherein the audio stream includes sinusoidal codes representing the frequency and the transient. According to the invention, the modified period of overlap (m) depends on the frequency value (f).

Audio coding

Techniques utilising Time Scale Modification (TSM) of signals are described. The signal is analysed and divided into frames of similar signal types. Techniques specific to the signal type are then applied to the frames thereby optimising the modification process. The method of the present invention enables TSM of different audio signal parts to be realized using different methods, and a system for effecting said method is also described.

Time-scale modification of signals

Wideband extension of telephone speech for higher perceptual quality. A method for extending the frequency range of a speech signal using wideband extension method with an inverse filter and a synthesis filter where both filters receive LPC coefficients from an LPC estimator. The wideband LPC coefficients are obtained from wideband LSFs. The wideband LSFs are obtained by appending highband LSFs, created by applying a matrix to narrowband LSFs, and lowband LSFs, created by dividing the narrowband LSFs by two. The matrix used to create the highband LSFs is selected from a predetermined list of matrices. The selection is based on either wideband or narrowband reflection coefficients extracted from the narrowband speech signal.

Wideband extension of telephone speech for higher perceptual quality

Described is a transmission system () comprising a transmitter () for transmitting a narrowband audio signal to a receiver () via a transmission channel (). The receiver () comprises a frequency domain bandwidth extender () for extending a bandwidth of the received narrowband audio signal by complementing the received narrowband audio signal with a highband extension thereof. The bandwidth extender () comprises an amplitude extender () for extending the bandwidth of an amplitude spectrum of the received narrowband audio signal by mapping narrowband amplitudes onto highband amplitudes. The bandwidth extender () further comprises a phase extender () for extending the bandwidth of a phase spectrum of the received narrowband signal and a combiner () for combining the extended amplitude spectrum and the extended phase spectrum into a bandwidth extended audio signal. The transmission system () is characterized in that the amplitude extender () comprises an amplitude mapper () and first and second frequency scale transformers (). The first frequency scale transformer. () is arranged for transforming a linear frequency scale of the amplitude spectrum into a logarithmic frequency scale, e.g. the Bark scale. The amplitude mapper () is arranged for mapping according to the logarithmic frequency scale the narrowband amplitudes onto the highband amplitudes. The second frequency scale transformer () is arranged for transforming the logarithmic frequency scale of the extended amplitude spectrum into the linear frequency scale.

Wideband signal transmission system

Coding of an audio signal is provided where an indicator of the frequency variation of sinusoidal components of the signal is used in the tracking algorithm of a sinusoidal coder where sinusoidal parameters from appropriate sinusoids from consecutive segments are linked. By applying an indicator such as a warp factor or polynomial fitting, more accurate tracks are obtained. As a result, the sinusoids can be encoded more efficiently. Furthermore, a better audio quality can be obtained by improved phase continuation.

Audio coding based on frequency variations of sinusoidal components

A communication system () comprises a transmitter (), a receiver (), and an up/down link communication channel () arranged for data communication from the transmitter () through the up link communication channel () to the receiver (). The communication system () is further arranged to feedback data from the receiver () through the down link communication channel () to the transmitter (). The receiver () comprises a bad frame indicator () for providing a bad frame indication (BFI) upon receipt of a corrupted frame, which is present in synchronized data communicated over the up link communication channel (); and the transmitter () comprises resynchronization means () coupled to the down link communication channel () for receiving BFI related data and in response thereto recommencing data communication over the up link communication channel (), in accordance with a resynchronization procedure, which starts from a predetermined state. A fast acting feedback resynchronization procedure for a GSM speech system is presented which prevents substantial error propagation from occurring at the receiver end.

Communication system having bad frame indicator means for resynchronization purposes

In an audio transmission system, an input signal is split up into two spectral portions in a transmitter. These spectral portions are coded by their own respective coder. The low-frequency signal portion is coded by a regular narrow-band coder and the high frequency portion is coded using a coder that outputs LPC codes and signal amplitude codes. In the receiver, the low frequency signal portion is reconstructed by a narrow-band decoder and the high frequency portion is reconstructed by applying a high pass filter to a white noise signal and applying an LPC filter that is controlled by the LPC codes to this filtered white noise signal and adjusting the signal amplitude with an amplifier that is controlled using the amplitude codes of the transmitter. The reconstructed low frequency signal and the reconstructed high frequency signal are then combined to yield a reconstructed output signal containing both frequency ranges.

High frequency and low frequency audio signal encoding and decoding system

A transmission system comprising a transmitter with a signal encoder, the signal encoder having an input for a signal to be encoded and a codebook entry selector for selecting a codebook entry for obtaining a synthetic signal giving a best approximation of a signal representative of the input signal, wherein the codebook entry includes a plurality of samples that can assume more than two values and is identified with a sequence of symbols, a receiver having a decoder with a codebook for deriving the codebook entry from the sequence of symbols received from the transmitter, wherein the codebook entries corresponding to sequences of symbols that differ in one particular symbol value also differ in one single sample value.

Growing community of inventors

Eindhoven, Netherlands

Andreas Johannes Gerrits