摘要:
The present application provides an acoustic echo mitigation apparatus and method, an audio processing apparatus and a voice communication terminal. According to an embodiment, an acoustic echo mitigation apparatus is provided, including: an acoustic echo canceller for cancelling estimated acoustic echo from a microphone signal and outputting an error signal; a residual echo estimator for estimating residual echo power; and an acoustic echo suppressor for further suppressing residual echo and noise in the error signal based on the residual echo power and noise power. Here, the residual echo estimator is configured to be continuously adaptive to power change in the error signal. According to the embodiments of the present application, the acoustic echo mitigation apparatus and method can, at least, be well adaptive to the change of power of the error signal after the AEC processing, such as that caused by change of double-talk status, echo path properties, noise level and etc.
摘要:
Dynamische Störgeräusche sollen besser geschätzt werden können. Hierzu wird eine Vorrichtung und ein Verfahren zum Schätzen eines Störgeräusches durch Bereitstellen eines Werts (X (e jΩ )) für die Leistungsdichte eines Gesamtsignals, das ein Nutzsignal und das zu schätzende Störgeräusch enthält, in einem aktuellen Zeitfenster, Vergleichen des Werts des Gesamtsignals mit einem mit einem Verstärkungsfaktor (1 + ε) multiplizierten Schätzwert eines Störgeräusches aus einem dem aktuellen Zeitfenster vorausgehenden Zeitfenster (19) und Verwenden des kleineren (17) der beiden Werte des Vergleichs als Vorschätzwert für das Störgeräusch in dem aktuellen Zeitfenster. Außerdem wird ein Codebuchschätzwert ( Ŝ nnCB (e jΩ )) für das Störgeräusch in dem aktuellen Zeitfenster bereitgestellt. Schließlich wird der größere Wert (27) von dem Vorschätzwert und dem Codebuchschätzwert als Schätzwert ( Ŝ nn (e jΩ )) für das Störgeräusch in dem aktuellen Zeitfenster verwendet.
摘要:
Methods and apparatus for echo cancellation in a system having a speaker and a microphone are disclosed. The speaker receives a speaker signal x(t). The microphone receives a microphone signal d(t) containing a local signal s(t) and an echo signal x1(t) that is dependent on the speaker signal x(t). The microphone signal d(t) is filtered in parallel with first and second adaptive filters having complementary echo cancellation properties relative to each other. A minimum echo output e3(t) is determined from an output e1(t) of the first adaptive filter and an output e2(t) of the second adaptive filter. The minimum echo output has a smaller energy and less correlation to the speaker signal x(t). A microphone output is then generated using the minimum echo output e3(t).
摘要:
The invention intends to successively extract a proper speech zone from a speech inputted in such a fashion that noise is mixed in a speech to be recognized, and to remove noise from the detected speech zone. To this end, a noise position is estimated from an input waveform, a speech zone is detected from a speech inputted subsequently by using power information of a speech at the estimated noise position, and noise is removed from the speech in the detected speech zone by using spectrum information of the speech at the estimated noise position. Further, the estimated noise zone is updated as appropriate by using a result of comparison between the power information of the input speech and the power information of the speech in the estimated noise zone so that the noise position is always properly estimated.
摘要:
An audio signal is encoded. The signal is first divided into bands (600) for each a band a "yardstick" signal element is selected (608), and its quantized magnitude used for allocating lists, with less accuracy for quanitzing non-yardstick signal elements (624) the encoded signal is later decoded (918, 926).
摘要:
Un signal audio est codé. Le signal est d'abord divisé en bandes (600), et pour chaque bande un élément de signal "mètre pliant" est sélectionné (608), et son intensité quantifiée est utilisée à l'affectation de bits, la précision de quantification d'éléments de signaux "non mètre pliant" (624) étant moindre. Le signal codé est ensuite décodé (918, 926).
摘要:
A play-out device is provided for playing out an audio signal via a speaker to provide a sound signal, and a recording device for recording the sound signal to obtain a recorded signal comprising a recording of at least the sound signal. The play-out device is configured for generating noise suppression data comprising the audio signal, or a reference thereto, and timing information for enabling the audio signal to be correlated in time with the recorded signal. A noise suppression subsystem is provided with the recorded signal and the noise suppression data. The noise suppression subsystem comprises a timing manager for synchronizing the audio signal with the recorded signal based on the timing information, and a noise suppressor for processing the recorded signal based on said synchronized audio signal to obtain a processed signal in which the recording of the sound signal is suppressed. The noise suppression subsystem is thus enabled to perform noise suppression, even when not comprised in the play-out device but rather in another device such as the recording device.
摘要:
The invention provides a method for attenuating noise in an input signal, the method comprising the steps of: receiving the input signal, estimating the power of the input signal to obtain an input power estimate, determining a noise power value based on the input power estimate, the noise power value corresponding to an estimate of the noise power within the input signal, determining an attenuation factor based on the noise power value, and attenuating the input signal using the attenuation factor.
摘要:
The invention includes a method, apparatus, and computer program to selectively suppress wind noise while preserving narrow-band signals in acoustic data. Sound from one or several microphones is digitized into binary data. A time-frequency transform is applied to the data to produce a series of spectra. The spectra are analyzed to detect the presence of wind noise and narrow band signals. Wind noise is selectively suppressed while preserving the narrow band signals. The narrow band signal is interpolated through the times and frequencies when it is masked by the wind noise. A time series is then synthesized from the signal spectral estimate that can be listened to. This invention overcomes prior art limitations that require more than one microphone and an independent measurement of wind speed. Its application results in good-quality speech from data severely degraded by wind noise.