摘要:
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.
摘要:
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.
摘要:
Described is a transmission system (10) comprising a transmitter (12) for transmitting a narrowband audio signal to a receiver (14) via a transmission channel (16). The receiver (14) comprises a frequency domain bandwidth extender (18) 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 (18) comprises an amplitude extender (24) for extending the bandwidth of an amplitude spectrum of the received narrowband audio signal by mapping narrowband amplitudes onto highband amplitudes. The bandwidth extender (18) further comprises a phase extender (26) for extending the bandwidth of a phase spectrum of the received narrowband signal and a combiner (28) for combining the extended amplitude spectrum and the extended phase spectrum into a bandwidth extended audio signal. The transmission system (10) is characterized in that the amplitude extender (24) comprises an amplitude mapper (42) and first and second frequency scale transformers (40,44). The first frequency scale transformer. (40) 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 (42) is arranged for mapping according to the logarithmic frequency scale the narrowband amplitudes onto the highband amplitudes. The second frequency scale transformer (44) is arranged for transforming the logarithmic frequency scale of the extended amplitude spectrum into the linear frequency scale.
摘要:
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).
摘要:
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.
摘要:
A communication system (1) comprises a transmitter (2), a receiver (3), and an up/down link communication channel (4, 6) arranged for data communication from the transmitter (2) through the up link communication channel (4) to the receiver (3). The communication system (1) is further arranged to feedback data from the receiver (3) through the down link communication channel (6) to the transmitter (2). The receiver (3) comprises a bad frame indicator (5) 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 (4); and the transmitter (2) comprises resynchronization means (7) coupled to the down link communication channel (6) for receiving BFI related data and in response thereto recommencing data communication over the up link communication channel (4), 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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.