173 results about "Monaural" patented technology

In binaural stereo coding, only one monaural channel is encoded. An additional layer holds the parameters to retrieve the left and right signal. An encoder is disclosed which links transient information extracted from the mono encoded signal to parametric multi-channel layers to provide increased performance. Transient positions can either be directly derived from the bit-stream or be estimated from other encoded parameters (e.g. window-switching flag in mp3).

Techniques of making a recording of or transmitting a sound field from either multiple monaural or directional sound signals that reproduce through multiple discrete loud speakers a sound field with spatial harmonics that substantially exactly match those of the original sound field. Monaural sound sources are positioned during mastering to use contributions of all speaker channels in order to preserve the spatial harmonics. If a particular arrangement of speakers is different than what is assumed during mastering, the speaker signals are rematrixed at the home, theater or other sound reproduction location so that the spatial harmonics of the sound field reproduced by the different speaker arrangement match those of the original sound field. An alternative includes recording or transmitting directional microphone signals, or their spatial harmonic components, and then matrixing these signals at the sound reproduction location in a manner that takes into account the specific speaker arrangement. The techniques are described for both a two dimensional sound field and the more general three dimensional case, the latter based upon using spherical harmonics.

A parametric representation of a multi-channel audio signal describes the spatial properties of the audio signal well with compact side information when a coherence information, describing the coherence between a first and a second channel, is derived within a hierarchical encoding process only for channel pairs including a first channel having only information of a left side with respect to a listening position and including a second channel having only information from a right side with respect to a listening position. As within the hierarchical process the multiple audio channels of the audio signal are downmixed iteratively into monophonic channels, one can pick the relevant parameters from an encoding step involving only channel pairs carrying the information needed to describe the spatial properties of the multi-channel audio signal.

The present technology provides a robust noise suppression system which may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. An acoustic signal may be received and transformed to cochlear domain sub-band signals. Features such as pitch may be identified and tracked within the sub-band signals. Initial speech and noise models may be then be estimated at least in part from a probability analysis based on the tracked pitch sources. Speech and noise models may be resolved from the initial speech and noise models and noise reduction may be performed on the sub-band signals and an acoustic signal may be reconstructed from the noise-reduced sub-band signals.

A wireless earphone device capable of providing balanced stereo. The wireless headset includes a Bluetooth earphone device, an audio adaptor, and a stereo earphone device for giving balanced sounds on both left and right channels. When the Bluetooth earphone device and the stereo earphone device are connected with the audio adaptor respectively, an audio switching module of the Bluetooth earphone device determines to transmit stereo audio signal with first channel signal and second channel signal to the stereo earphone device through the audio adaptor. Otherwise, only one channel of the stereo audio signal is transmitted from the audio switching module to a mono speaker unit on the Bluetooth earphone device.

A parametric stereo upmix apparatus (300, 400) generating a left signal (206) and a right signal (207) from a mono downmix signal (204) based on spatial parameters (205). Said parametric stereo upmix being characterized in that it comprises a means (310) for predicting a difference signal (311) comprising a difference between the left signal (206) and the right signal (207) based on the mono downmix signal (204) scaled with a prediction coefficient (321). Said prediction coefficient is derived from the spatial parameters (205). Said parametric stereo upmix apparatus (300, 400) further comprises an arithmetic means (330) for deriving the left signal (206) and the right signal (207) based on a sum and a difference of the mono downmix signal (204) and said difference signal (311).

Provided are a speech recognition training processing method and an apparatus including the same. The speech recognition training processing method includes acquiring multi-talker mixed speech sequence data corresponding to a plurality of speakers, encoding the multi-speaker mixed speech sequence data into an embedded sequence data, generating speaker specific context vectors at each frame based on the embedded sequence, generating senone posteriors for each of the speaker based on the speaker specific context vectors and updating an acoustic model by performing permutation invariant training (PIT) model training based on the senone posteriors.

Two digital data sets are combined by equating a first subset of samples to neighboring samples from a second subset which is interleaved with the first subset of samples where the equated samples of the two digital data sets do not correspond in time, and by subsequently adding corresponding samples from both digital data sets. This results in a third digital data set that allows the unraveling of the two digital data sets. The third digital data set, when combining two digital audio streams into a single digital audio stream, is still a good mono representation of the two combined digital audio streams and can thus be reproduced on regular reproduction equipment, yet the use of a decoder according to the invention allows the unraveling of the two digital data sets from the third digital data set

This invention provides a system capable of transitioning from surround sound effect to stereo sound effect, and then to a monophonic sound effect using front speakers. A user may control the amount of surround sound effect that is combined with stereo sound effect, and the amount of stereo sound effect that is combined with monophonic sound effect. The user may also control the system to hear pure surround sound effect, stereo sound effect, or monophonic sound effect. The system may allow a user to control the contribution of a particular sound effect by controlling the relative proportions of the filtered dipole signals and the attenuated unfiltered dipole signals. The signal processing may be also done through an analog device. The system may also process some audio channels in an inverted form and compensate for these inversions by reversing the polarity of that output to the speaker, as well as accept two-channel and four-channel audio inputs.

A stereo audio encoder (100) comprises a parametric stereo encoder (115) which generates a mono signal and parametric stereo parameters for at least a high frequency part of an input stereo signal. A stereo intensity encoder (117) generates stereo intensity data for the mono signal. The mono signal and intensity data are encoded in accordance with an encoding standard such as MPEG Layer II and the parametric stereo parameters are included in the ancillary data sections by an output processor (113). Thus, a legacy decoder (such as an MPEG Layer II decoder) may generate a stereo signal using the stereo intensity data whereas a higher complexity decoder may generate a high quality audio signal using the parametric stereo parameters. A stereo decoder (200) receives the encoded data from the encoder (100). An intensity decoder (203) generates a stereo signal using intensity data. This is fed to a parametric stereo decoder (207) which processes the stereo signal in accordance with extracted parametric stereo data.

Disclosed is a method for generating a surround-channel audio signal (Mout) from a mono/stereo audio signal (Min, Sin), comprising the steps of: a) generating a first multi-channel signal (M1) by surround panning the mono/stereo audio signal (Sin); b) generating a second multi-channel signal (M2) by effect processing the mono/stereo input signal (Min, Sin) so that the rear signals comprise at least reverberation of the mono/stereo audio signals; and c) mixing the corresponding signals of the first multi-channel signal (M1) and the second multi-channel signal (M2), thereby forming the surround-channel audio signal (Mout).

A pair of matched monaural speakers are added inside conventional left and right stereo speaker housings so that the left, right and flat panel monitor appear joined in a seamless aesthetic effect. The added speakers are provided with center channel monaural sound, making it appear as though such sound emanates from the center of the monitor, thereby enhancing home theater impact to provide stereo and center channel sound from each left and right speaker housing.

A USB communication device using an earphone jack of a mobile communication terminal and a method for judging the type of the external device when the external device is connected to the earphone jack of the mobile communication terminal. When the external device is a USB external camera, the mobile communication terminal receives the converted image from the USB external camera through the earphone jack; The headphone earpiece of the headphone microphone chain, and receive mono sound from the microphone of the headphone microphone chain through the microphone signal line of the headphone jack. Therefore, in the present invention, the functions of USB communication between the mobile communication terminal and the external device, stereo output and monaural output can be selectively performed according to the kind of the external device connected to the earphone jack.

The invention discloses a speaker-independent single-channel voice separation method, which comprises the following steps: preparing a data set, and carrying out data preprocessing; establishing a single-channel voice separation model based on complex ideal floating value masking; when the single-channel voice separation model is trained, adopting statement-level replacement invariance training; and inputting the mixed voice data into the trained model for voice separation. According to the speaker-independent single-channel voice separation method based on statement-level replacement invariance training and complex ideal floating value masking, complex ideal floating value masking estimation is effectively and accurately realized through statement-level replacement invariant training; inthe method, a bidirectional long-short-term memory neural network structure is adopted to estimate complex number ideal floating value masking, and a statement-level replacement invariant training standard is further utilized to solve the problem of label blurring, so that single-channel voice separation has a good effect.

A stereo headphone employing a standard stereo headphone plug is adapted for automatically hearing a monaural signal at both earpieces, when accessing a typical monaural source. In the first embodiment of the invention, an impedance element couples the signal from a first acoustical driver that receives the monaural signal from the stereo plug tip, to a second acoustical driver that is connected to the stereo plug ring and normally receives no signal when plugged into a conventional monaural audio source output jack. The magnitude of the coupling impedance is selected with respect to the impedance of the acoustical driver so that the reduction in loudness at the second earpiece due to the signal voltage drop across the coupling impedance is not perceptible to the listener. This will occur when the reduction in loudness at the second earpiece is less than the threshold of perceivable loudness reduction at one ear when there is no reduction in loudness at the other ear. The effect of the coupling impedance, when listening to a stereo audio source is insignificant, firstly, because the two stereo channel signals appear at their respective drivers with virtually no attenuation due to the coupling impedance. Secondly, although the coupling impedance does contribute a slight amount of additional crosstalk between the stereo channels, the magnitude of the increase in crosstalk is dependent upon the ratio of the coupling impedance to the output impedance of the stereo source. A typical stereo source for which the use of this headphone is intended has an output impedance so low compared to the coupling impedance that the increase in crosstalk is too small to be perceptible as affecting the stereo separation or the stereo imaging afforded by the stereo source. In the second embodiment of the invention two equal impedance elements couple the monaural signal from the stereo plug tip to each acoustical driver. This equalizes the loudness of the monaural signal heard at each earpiece and slightly reduces the level of one stereo channel with respect to the other. Crosstalk and stereo imaging are virtually unaffected, as with the first embodiment.

The invention provides a digital signal processor (DSP) implementation system for a two-channel convolution mixed voice signal blind source separation algorithm. The system comprises an embedded processor platform, a microphone access circuit, a power circuit and a signal pre-amplification circuit, wherein the embedded processor platform is used for finishing analogue/digital (A/D) conversion, D/A conversion and analogue signal output; the microphone access circuit comprises two single-channel microphones for acquiring voice signals and a two-channel microphone acquisition circuit; two-channel voice signals acquired by the two single-channel microphones are taken as the left and right channel signal input of a line in audio interface; the signal pre-amplification circuit effectively amplifies the signals of the microphones until gain required by the line in audio interface is achieved. A plurality of functions of microphone acquisition, pre-amplification, voice blind source separation, voice play and storage and the like of the two-channel voice signals of two speakers are realized, and the actually acquired mixed voices of the two speakers can be effectively separated under both an offline condition and an online condition.

The embodiment of the invention provides a device, a system and a method for coding and decoding, which are applied to a stereo signal. The method comprising the following steps: acquiring a stereo parameter of the stereo signal to be coded, which is converted in an MDCT field, and performing quantification processing on the stereo parameter; downwards mixing the stereo signal into a signal-channel signal code; compounding and sending the quantified stereo parameter and a coded signal; decoding and compounding the data to be decoded into the stereo parameter and the coded signal; decoding andquantifying the stereo parameter; decoding the coded signal into a single-channel signal; acquiring decorrelation signals of the single-channel signal in the MDCT field; and synthesizing and restoring the quantified stereo parameter, the single-channel signal and the decorrelation signals to the stereo signal. The device, the system and the method for coding and decoding are used for coding stereo and can solve the problem how to obtain two paths of left and right channel stereo signals by one path of signal-channel signal at a decoding end and realize stereo coding and decoding in the MDCT field through obtaining the decorrelation signals in the MDCT field.

The present invention provides significant and much needed improvements in person-to-person interactions within videoconferencing events, by enabling each participant a greater freedom the move around quickly, and change their acoustic perspectives based on their relative positions to other participants, to control who is near to them, and to quickly scan through all participants in a venue to choose participants they like and later to see who liked them. Recent advances in processing power and simulation of binaural sound from monophonic sound simulates distances and azimuth angles as well as aural anatomical comb filtering effects to simulated stereophonic sound and select desirable crowd noises in videoconferencing.

A scalable encoding device for realizing scalable encoding by CELP encoding of a stereo sound signal and improving the encoding efficiency. In this device, an adder and a multiplier obtain an average of a first channel signal CH1 and a second channel signal CH2 as a monaural signal M. A CELP encoder for a monaural signal subjects the monaural signal M to CELP encoding, outputs the obtained encoded parameter to outside, and outputs a synthesized monaural signal M′ synthesized by using the encoded parameter to a first channel signal encoder. By using the synthesized monaural signal M′ and the second channel signal CH2, the first channel signal encoder subjects the first channel signal CH1 to CELP encoding to minimize the sum of the encoding distortion of the first channel signal CH1 and the encoding distortion of the second channel signal CH2.