摘要:
Provided is an audio system in which an audio signal to be heard can be automatically set to an appropriate volume or sound quality. The audio system is provided with: a mobile terminal device provided with a microphone unit that collects surrounding sound waves, and a transmission unit that, on the basis of the sound waves collected by the microphone unit, sets the volume or sound quality of an audio signal to be amplified by an audio device, and transmits the set audio signal volume or sound quality information to the audio device; and an audio device provided with an amplification unit that amplifies the audio signal, a control unit that adjusts the audio signal volume or sound quality by means of the amplifier, and a receiving unit that receives the audio signal volume or sound quality setting information from the mobile terminal device. The control unit controls the amplification unit so as to adjust the audio signal volume or sound quality on the basis of the audio signal volume or sound quality setting information received by the receiving unit.
摘要:
According to some embodiments of the present invention there is provided a method of using an earphone output speaker as a microphone for a phone call between two and/or more participants, or for measuring biometric data of a user. The method may comprise playing a received signal to an electro-acoustic output transducer of an earphone. The method may comprise instructing an audio processing circuit of a local client terminal to record an audio signal from the same electro-acoustic output transducer. The method may comprise calculating a voice signal and/or a biometric measurement based on a function combining the recorded audio signal, the received signal, and filtration coefficients, using a processing unit of the local client terminal. The method may comprise sending the voice signal and/or a biometric measurement through an output interface of the local client terminal.
摘要:
A howling removing apparatus according to the present disclosure is a howling removing apparatus to be connected to a microphone and a speaker. The howling removing apparatus includes: a nonlinear converter that nonlinearly converts a sound signal input to the speaker and outputs a nonlinear signal; a delay unit that delays the sound signal by a fixed time and outputs a delay signal; a norm calculator that calculates a norm from the delay signal; a filter coefficient generator that, based on the nonlinear signal, the delay signal and the norm, generates an adaptive filter that simulates a transfer characteristic of a space where the sound signal is reproduced from the speaker and is returned to the microphone; a cancel signal generator that convolves the delay signal and the adaptive filter with each other and generates a cancel signal; and a subtracter that subtracts the cancel signal from the sound signal. When an average sound pressure level of the sound signal exceeds a threshold value, the filter coefficient generator initializes the adaptive filter, thereby sensing oscillation of the adaptive filter even when a positional relationship between the microphone and the speaker is dynamically changed, and returning an output to normal.
摘要:
A method, and one or more non-transitory computer-readable media storing instructions, and a device for removal of unwanted components in an audio signal, the device comprising a processor, coupled to memory, configured to receive reference and processed inputs into memory where the processed input is a result of a reduction process of unwanted components of the audio signal, estimate envelope values for processed and reference inputs at a plurality of time and frequency instances, for each time and frequency instance: compute a first gain in relation to a ratio of the estimated envelope value of the processed input to the estimated envelope value of the reference input, apply a nonlinear process to said first gain to produce a second gain, compute an output gain as the ratio between second gain and first gain and, apply the output gain to processed input, thereby producing a filtered output with unwanted components suppressed.
摘要:
A method and system for improved audio quality in teleconferencing are provided. The method includes analyzing the audio signal of multiple input lines in a teleconferencing system to detect if any two input lines contain substantially the same audio signal with a delay shorter than that of a conventional echo caused by an input line's own audio feedback via a teleconferencing server. The method further includes selecting the input line with the higher amplitude audio signal or the earlier received audio signal when two input lines with substantially the same audio signal are detected.
摘要:
Provided are systems and methods for contextual switching of microphones in audio devices. An example method includes detecting a change in conditions for capturing an acoustic signal by at least two microphones. A configuration is associated with the at least two microphones. The example method provides determining that the change in conditions has been stable for a pre-determined period of time. In response to the determination, the method changes the configuration associated with the at least two microphones. The conditions may include an absence or presence of far-end speech, reverberation, low/high signal-to-noise ratio, low/high signal-to-echo ratio, type of background noise, and so on. The changing of the configuration includes assigning a primary microphone and a secondary microphone based on a change in conditions. Based on the changed configuration, tuning parameters may be adjusted for noise suppression and acoustic echo cancellation.
摘要:
A communication device and method including a microphone, a first filter coupled to the microphone adapted to pass signals in an audio band only, a second filter coupled to the microphone adapted to pass signals only in an upper region of the audio band, an optional amplifier or two, an analog to digital converter (ADC) unit or two, an optional a switching unit connected between an input of the ADC unit and an output of each of the first and second filters and configured to selecting between the first and second filters, and a controller configured to control the switching unit and/or to select the second filter for acoustic communication, where the acoustic communication is adapted to the upper region of the audio band.
摘要:
To address issues with present echo gate control, a method and apparatus for more intelligently operating an echo gate is described herein. In particular, the decision of whether to mute an uplink signal, or not, is formulated herein as primarily a perceptual decision based on an appropriate analysis of the perceptual interaction of the current residual echo and the current near-end signal(s). By doing so, the application of muting through an echo gate may be minimized and/or more appropriately engaged. This will lead to fewer dropouts and muting of speech onsets and offsets 1) during periods such as double-talk or 2) during periods of downlink playback in the presence of low near-end signal levels, two cases of particular importance.
摘要:
Systems and methods are provided for echo reduction. For example, first audio data associated with a current time unit is obtained, the first audio data being generated by an audio-acquisition device of a receiver based on at least information associated with a first audio signal acquired by the audio-acquisition device and a second audio signal output from an audio-playback device of the receiver, the second audio signal being output from the audio-playback device of the receiver based on at least information associated with second audio data transmitted previously by a sender before the current time unit; third audio data transmitted by the sender during a second time unit preceding the current time unit is obtained; and fourth audio data is obtained by performing echo reduction on the first audio data based on at least information associated with the third audio data.
摘要:
Provided are systems and methods for contextual switching of microphones in audio devices. An example method includes detecting a change in conditions for capturing an acoustic signal by at least two microphones. A configuration is associated with the at least two microphones. The example method provides determining that the change in conditions has been stable for a pre-determined period of time. In response to the determination, the method changes the configuration associated with the at least two microphones. The conditions may include an absence or presence of far-end speech, reverberation, low/high signal-to-noise ratio, low/high signal-to-echo ratio, type of background noise, and so on. The changing of the configuration includes assigning a primary microphone and a secondary microphone based on a change in conditions. Based on the changed configuration, tuning parameters may be adjusted for noise suppression and acoustic echo cancellation.