Abstract: A system includes a first integrated circuit device, a second integrated circuit device, and a reference clock provided to the first and second integrated circuit devices. The first integrated circuit device detects a first edge of a first clock utilized by the first integrated circuit device, detects a second edge of the first clock, determines a first count of cycles of the reference clock between the first edge and the second edge, and communicates the first count to the second integrated circuit device. The second integrated circuit device receives the first count, provides a third edge of a second clock utilized by the second integrated circuit device, determines that a first number of cycles of the reference clock since providing the third edge is equal to the first count, and provides a fourth edge of the second clock in response to determining that the first number of cycles is equal to the first count.

Abstract: A system (100) comprising: a first unit (104) and one or more second units (104). The first unit (102) comprises: a timing reference (114) configured to provide a master-timing-reference-signal; a master time block configured to provide a master-time-signal (117) for the first unit (102) based on the master-timing-reference-signal; and a first interface (122) configured to: receive timestamped-processed-second-RF-signals from the one or more second units (104); and provide a first-unit-timing-signal (262) to the one or more second units (104) based on the master-time-signal.

Abstract: A system (100) comprising: a first unit (104) and one or more second units (104). The first unit (102) comprises: a timing reference (114) configured to provide a master-timing-reference-signal; a master time block configured to provide a master-time-signal (117) for the first unit (102) based on the master-timing-reference-signal; and a first interface (122) configured to: receive timestamped-processed-second-RF-signals from the one or more second units (104); and provide a first-unit-timing-signal (262) to the one or more second units (104) based on the master-time-signal.

Abstract: A device (1) 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 (2) 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 (3) for synthesizing the noise components using the noise parameters of the selected sets only.

Abstract: A decoder receives (501) a bitstream comprising an encoded mono signal and stereo data. A time scale processor (503) 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 (509) generates a stereo signal for the frequency sample blocks and these are converted to the time domain by a frequency-to-time processor (511). A synchronization processor (515) 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.

Abstract: 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 (Cs) comprising a representation level (r) for each frame or including sinusoidal codes (Cs) where some of these codes comprise a phase (?), a frequency (?) 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.

Abstract: 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 (40) to determine one or more sinusoidal components for each of the plurality of sequential segments. The sinusoidal components are linked (42) 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 (46).

Abstract: 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.

Abstract: 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.

Abstract: A device (1) for producing sound samples from sound parameters representing sound components comprises a transient synthesis unit (14) for synthesizing transient sound components from transient sound parameters contained in each frame. To increase the efficiency of the synthesis, a transient selection unit (11) is arranged for selecting only a single transient sound component per frame. Additionally, the device may be arranged for producing fewer sinusoidal sound components if a transient is produced. Transform domain coefficients may be convolved with a transform domain representation of a time window representation, the number of resulting transform domain coefficients being controlled to further enhance the efficiency of the synthesis.

Abstract: 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.

Abstract: 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, W2, W3), and consecutive segments have a normal period of overlap (0) of their trailing edges and leading edges. Segments in which a transient5 component 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 (W2m) having a modified leading edge, so that the modified trailing edge and the modified leading edge have a modified period of overlap (0m) that comprises the transient component and that is shorter than the normal period of overlap (0), and wherein the audio stream includes sinusoidal codes representing the frequency and the transient. According to the invention, the modified period of overlap (0m) depends on the frequency value (f).

Abstract: A decoder receives (501) a bitstream comprising an encoded mono signal and stereo data. A time scale processor (503) 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 (509) generates a stereo signal for the frequency sample blocks and these are converted to the time domain by a frequency-to-time processor (511). A synchronization processor (515) 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.

Abstract: A hybrid sinusoidal/pulse excitation encoder has been recently proposed for constructing a scalable audio encoder The base layer consisting of data supplied by the sinusoidal encoder retains the main features of the input signal achieving medium to high quality audio at a very low bit rate. Quality can be further enhanced by adding excitation signal layers associated with a decreasing decimation that increasingly model more subtle aspects of the original signal. The invention provides a method of mixing the different excitation signal layers so that the full concept of scalability is realised without compromising the quality of the encoded signals. The mixing is controlled via a quality parameter that weights the significance of previous layers when constructing a new higher layer.

Abstract: The invention relates to an audio encoder and decoder and methods for audio encoding and decoding. In the encoder an audio signal is split into an anechoic signal part and information regarding a reverberant field associated with the audio signal, preferably by a representation using only few parameters such as reverberation time and reverberation amplitude. The anechoic signal is then encoded using an audio codec. At the decoder the anechoic signal part is restored using the audio codec, and the restored anechoic signal part is transformed into the substantially original audio signal by applying reverberance according to the information regarding the reverberant field, preferably by convolution with a room impulse response generated on the basis of the reverberant field information. According to the invention the audio codec involved needs only be capable of encoding anechoic audio signals, thus solving the problem of parametric audio codecs providing poor performance on reverberant audio signals.

Abstract: A method of encoding a digital audio signal, wherein for each time segment the signal is spectrally flattened to obtain a spectrally flattened signal (r) and possibly spectral flattening parameters (LPP). The spectrally flattened signal is modelled by an excitation signal comprising a first partial excitation signal (px) conforming to an excitation signal generated by an RPE or CELP technique, and a second partial excitation signal (PEp) being a set of extra pulses with arbitrary positions and amplitudes. An audio bit stream as comprising the first and second partial excitation signals is generated. The extra pulses can be added to the excitation signal at positions in time that correspond to the time of occurrence of the spike, or preferably at positions in time of an RPE time grid.

Abstract: 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 ERE based. The perceptual relevance value may involve the amplitude of the respective sinusoidal component, and/or the envelope of the respective channel.

Abstract: A sound decoding device (1) is arranged for decoding sound represented by sets of parameters, each set comprising sinusoidal parameters (SP) representing sinusoidal components of the sound and further parameters (NP, TP) representing further components of the sound, such as noise and/or transients. The device comprises a separate sinusoids generator unit (17, 18) for each output channel (L, R), while the further component generator units (20; 21) are shared between the channels.

Abstract: 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.

Abstract: A device (1) 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 (2) 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 (3) for synthesizing the noise components using the noise parameters of the selected sets only.