摘要:
A microphone arrangement and a method using the microphone arrangement for recording surround sound in a mobile device, where the microphone arrangement comprises a first and a second microphone and arranged at a first distance to each other and configured to obtain a stereo signal, and comprises a third microphone configured to obtain a steering signal together with at least one of the first and second microphone or with a fourth microphone. The microphone arrangement also comprises a processor configured to separate the stereo signal into a front stereo signal and a back stereo signal based on the steering signal.
摘要:
The invention relates to a method for rendering a stereo audio signal over a first loudspeaker and a second loudspeaker with respect to a desired direction, the stereo audio signal comprising a first audio signal component (L) and a second audio signal component (R), the method comprising: providing a first rendering signal based on a combination of L and a first difference signal obtained based on a difference between L and R to the first loudspeaker, and providing a second rendering signal based on a combination of R and a second difference signal obtained based on the difference between L and R to the second loudspeaker, such that both difference signals are different with respect to sign and one difference signal is delayed by a delay compared to the other difference signal to define a dipole signal, wherein the delay is adapted according to the desired direction.
摘要:
A virtual stereo synthesis method includes acquiring at least one sound input signal on a first side and at least one sound input signal on a second side, separately performing ratio processing on a preset head related transfer function (HRTF) left-ear component and a preset HRTF right-ear component of each sound input signal on the second side, to obtain a filtering function of each sound input signal on the second side, separately performing convolution filtering on each sound input signal on the second side and the filtering function of the sound input signal on the second side, to obtain the filtered signal on the second side, and synthesizing all of the sound input signals on the first side and all of the filtered signals on the second side into a virtual stereo signal where the method may alleviate a coloration effect, and reduce calculation complexity.
摘要:
The invention relates to a method for processing a multi-channel audio signal which carries a plurality of audio channel signals. The method comprises determining a time-scaling position using the plurality of audio channel signals and time-scaling each audio channel signal of the plurality of audio channel signals according to the time-scaling position to obtain a plurality of time scaled audio channel signals.
摘要:
The invention relates to a parametric audio encoder, comprising a parameter generator, the parameter generator being configured to determine a first set of encoding parameters and reference audio signal values, wherein the reference audio signal is another audio channel signal or a downmix audio signal derived from at least two audio channel signals of the plurality of multi-channel audio signals, to determine a first encoding parameter average based on the first set of encoding parameters of the audio channel signal, to determine a second encoding parameter average based on the first encoding parameter average of the audio channel signal and at least one other first encoding parameter average of the audio channel signal, and to determine the encoding parameter based on the first encoding parameter average of the audio channel signal and the second encoding parameter average of the audio channel signal.
摘要:
The invention relates to an audio signal synthesizer, the audio signal synthesizer comprises a transformer for transforming the down-mix audio signal into frequency domain to obtain a transformed audio signal; a signal generator for generating a first auxiliary signal, for generating a second auxiliary signal, and for generating a third auxiliary signal upon the basis of the transformed audio signal; a de-correlator for generating a first de-correlated signal, and for generating a second de-correlated signal from the third auxiliary signal, the first de-correlated signal and the second de-correlated signal being at least partly de-correlated; and a combiner for combining the first auxiliary signal with the first de-correlated signal to obtain a first audio signal, and for combining the second auxiliary signal with the second de-correlated signal to obtain the second audio signal, the first audio signal and the second audio signal forming the multi-channel audio signal.
摘要:
Methods and devices for a low complex inter-channel phase difference estimation are provided. A method for the estimation of inter-channel phase differences (IPDs), comprises applying a transformation from a time domain to a frequency domain to a plurality of audio channel signals, calculating a plurality of IPD values for the IPDs between at least one of the plurality of audio channel signals and a reference audio channel signal over a predetermined frequency range, each IPD value being calculated over a portion of the predetermined frequency range, calculating, for each of the plurality of IPD values, a weighted IPD value by multiplying each of the plurality of IPD values with a corresponding frequency-dependent weighting factor, and calculating an IPD range value for the predetermined frequency range by adding the plurality of weighted IPD values.
摘要:
According to the invention, a device for post-processing at least one channel signal of a plurality of channel signals of a multi-channel signal is described, the at least one channel signal being generated from a decoded downmix signal by a low-bit-rate audio coding/decoding system, the device comprising: a receiver for receiving the at least one channel signal generated from the decoded downmix signal, a time envelope of the decoded downmix signal, an interchannel time difference between the channel signal and the downmix signal, and a classification indication indicating a transient type of the downmix signal; and a post-processor for post-processing the at least one channel signal based on the time envelope of the decoded downmix signal weighted by a respective weighting factor and in dependence on the classification indication and the interchannel time difference.
摘要:
The invention relates to an audio signal synthesizer, the audio signal synthesizer comprises a transformer for transforming the down-mix audio signal into frequency domain to obtain a transformed audio signal; a signal generator for generating a first auxiliary signal, for generating a second auxiliary signal, and for generating a third auxiliary signal upon the basis of the transformed audio signal; a de-correlator for generating a first de-correlated signal, and for generating a second de-correlated signal from the third auxiliary signal, the first de-correlated signal and the second de-correlated signal being at least partly de-correlated; and a combiner for combining the first auxiliary signal with the first de-correlated signal to obtain a first audio signal, and for combining the second auxiliary signal with the second de-correlated signal to obtain the second audio signal, the first audio signal and the second audio signal forming the multi-channel audio signal.
摘要:
Embodiments of the present invention provide a method and an apparatus for generating a sideband residual signal. The method includes: comparing energy of a first signal input by a first sound channel with energy of a second signal input by a second sound channel; if the energy of the first signal is greater than the energy of the second signal, generating a sideband residual signal by allocating a monophonic quantization error to the first signal; and if the energy of the first signal is smaller than the energy of the second signal, generating a sideband residual signal by allocating a monophonic quantization error to the second signal. By using the method and apparatus provided in the embodiments of the present invention, it can be avoided that a monophonic quantization error has a greater impact on a signal whose energy is smaller.