The present invention relates to a method for audio processing of an audio signal captured by a microphone.
Microphones are used to capture an audio signal, in particular speech of a user. However, the microphone records both the speech signals of the user and ambient noise. However, for the most part this ambient noise is not desired in the captured audio signal.
In the priority application of this application, the German Patent and Trademark Office has searched the following documents: JP 2015-192256 A and US 2022/0201414 A1.
It is therefore an object of the present invention to provide a method for audio processing of an audio signal captured by a microphone, which method enables improved capture of a useful signal (e.g., a speech signal) in an environment in which ambient noise is present.
This object is achieved by a method for audio processing of an audio signal according to claim 1.
A method is thus provided for audio post-processing of an audio signal that has been captured by a microphone. In this case, reverberation properties of a room can first be determined. This is accomplished by capturing a first audio signal as a test signal by at least one microphone that is located in the room. The reverberation properties of the room are determined based on the captured first audio signals of the at least one microphone. A second audio signal can be captured by the microphone in the room. The second audio signal comprises a useful sound and a room reverberation generated by the useful sound in the room. A prediction of a room reverberation is created based on the determined reverberation properties of the room and the captured second audio signal. The second audio signal is analyzed for locations at which the room reverberation has a higher level than a useful signal, in particular a speech signal. A volume of a second audio signal to be output is reduced at those locations at which the prediction of the room reverberation indicates that the room reverberation exceeds a threshold value. In other words, the volume of the second audio signal to be output is reduced depending on the room reverberation. In particular, if the useful signal in the second audio signal has a low level whilst the room reverberation has a high level, then the volume of the second audio signal to be output can be reduced in order to reduce the influence of the room reverberation on the entire second audio signal to be output. Depending on the room reverberation, the volume or level of the second audio signal to be output is reduced. At points where the useful signal (e.g. the speech signal) has a higher level than the room reverberation, the volume does not have to be reduced because the level of the useful signal exceeds the level of the room reverberation.
The useful signal can optionally be a speech signal or an audio signal generated by a user (e.g. singing). The useful signal represents the desired portion of the captured audio signal.
According to one aspect of the present invention, a measurement of linear properties of the system consisting of the room and the microphone can be carried out. This can be accomplished, for example, by means of an impulse response. Based on these measurements, a reverberation time of the room in which the microphone is located can be determined.
According to one aspect of the present invention, an algorithm is provided which follows the envelope curve of the input signal and detects transients and their volume or level.
According to one aspect of the present invention, the volume of the second audio signal to be output is reduced if it has been captured based on the algorithm that room reverberation and only a small proportion of useful signal are present in the audio signal. In particular, the volume can be reduced if the signal level of the reverberation is below a threshold value compared to the transient level.
According to one aspect of the present invention, a volume reduction can be carried out taking into account parameters which are captured by the measurement. In particular, the reverberation time can represent the attack time. In this way, the acoustic properties of the room can be parameterized in a manner analogous to noise gate processing.
The invention also relates to an audio post-processing unit for audio post-processing of an audio signal that has been captured by a microphone. In this case, reverberation properties of a room can first be determined. This is accomplished by capturing a first audio signal as a test signal by at least one microphone that is located in the room. The reverberation properties of the room are determined based on the captured first audio signals of the at least one microphone. A second audio signal can be captured by the microphone in the room. The second audio signal comprises a useful sound and a room reverberation generated by the useful sound in the room. A prediction of a room reverberation is created based on the determined reverberation properties of the room and the captured second audio signal. The second audio signal is analyzed for locations at which the room reverberation has a higher level than a useful signal, in particular a speech signal. A volume of a second audio signal to be output is reduced at those locations at which the prediction of the room reverberation indicates that the room reverberation exceeds a threshold value. In other words, the volume of the second audio signal to be output is reduced depending on the room reverberation. In particular, if the useful signal in the second audio signal has a low level whilst the room reverberation has a high level, then the volume of the second audio signal to be output can be reduced in order to reduce the influence of the room reverberation on the entire second audio signal to be output.
Further embodiments of the invention are the subject of the dependent claims.
Advantages and exemplary embodiments of the invention are explained in more detail hereinafter with reference to the drawing.
To determine the room reverberation 120, a microphone (e.g. the microphone 200) can be placed in the room 100. An audio signal can then be output as a test signal in the room 100. The microphone 200 can then capture the test signal and the room reverberation generated thereby in order to be able to determine the room reverberation properties of the room 100.
The captured audio signal of the microphone 200 can be post-processed in an audio processing unit 300.
As long as the level of the room reverberation 120 is lower than the level of the useful sound 110, the influence of the reverberation sound 120 on the output signal is relatively small. Since the useful signal 110, for example a speech signal, does not occur uniformly, there may be points in the captured audio signal at which the useful sound 110 has a lower level than the room reverberation 120. In such cases, the room reverberation 120 then becomes noticeable in the captured audio signal. This is not desirable. In order to avoid this, it is proposed to reduce the level of the audio signal to be output (useful sound plus room reverberation) at those points at which the room reverberation 120 is above a threshold value. This is typically the case if the level of the useful sound 110 is lower than the level of the room reverberation 120 at this time. In these cases, the level of the audio signal (useful signal and room reverberation) can be reduced. This reduces the influence of the room reverberation 120 on the overall audio signal.
The audio processing unit 300 further comprises a gate 320 with a side input 321. The gate 320 is controlled depending on the signal at the side input 321. Optionally, the gate 320 can decouple the output 302 from the input 301 so that no output signal 302 is output.
Optionally, the gate 320 can reduce a level of the input signal 301 if the signal at the side input 321 indicates that a level of a room reverberation is higher than a level of a useful signal.
| Number | Date | Country | Kind |
|---|---|---|---|
| 102024100423.1 | Jan 2024 | DE | national |