A method of double talk detection includes using up-sampling. Audio signals received from the far end are up-sampled prior to output by the loudspeaker at the near end. The microphone at the near end captures audio at the up-sampled rate, and the audio output by the loudspeaker is detectable due to having no energy in the up-sampled frequency bands. The double talk detector uses this information to generate a signal for suppressing the echo of the far end audio from the captured audio signal that is transmitted to the far end.

A sound pick-up device includes an adaptive filter configured to generate, from a reference signal, an estimated noise signal indicating a component of a noise signal contained in an input signal, a noise elimination signal generator configured to generate a noise elimination signal by subtracting the estimated noise signal from the input signal, a filter coefficient update unit configured to update a filter coefficient of the adaptive filter using the noise elimination signal, and a sample position identification unit configured to identify at least one signal sample position among a plurality of signal sample positions from a signal sample position where the noise elimination signal has a maximum absolute value to a signal sample position where the noise elimination signal has an absolute value that is largest at a predetermined order, in which the filter coefficient update unit updates the filter coefficient at the at least one signal sample position.

An audio data processing method and apparatus, a device, and a storage medium that can obtain, in a game voice mode, a signal processing result associated with a first pre-signal processing policy in an application layer of a service application; control, in the application layer according to the signal processing result, a turned-on/turned-off state of a second optimization component in a second pre-signal processing policy in a terminal system layer, or a turned-on/turned-off state of a first optimization component in the first pre-signal processing policy, a first optimization component turned on in the first pre-signal processing policy being different from a second optimization component turned on in the second pre-signal processing policy; and obtain uplink voice data of a first user in the game voice mode, and perform voice optimization on the uplink voice data based on the turned-on first optimization component and the turned-on second optimization component.

Example techniques involve noise-robust acoustic echo cancellation. An example implementation may involve causing one or more speakers of the playback device to play back audio content and while the audio content is playing back, capturing, via the one or more microphones, audio within an acoustic environment that includes the audio playback. The example implementation may involve determining measured and reference signals in the STFT domain. During each n.sup.th iteration of an acoustic echo canceller (AEC): the implementation may involve determining a frame of an output signal by generating a frame of a model signal by passing a frame of the reference signal through an instance of an adaptive filter and then redacting the n.sup.th frame of the model signal from an n.sup.th frame of the measured signal. The implementation may further involve determining an instance of the adaptive filter for a next iteration of the AEC.

Systems, apparatus, and methods for processing audio signals associated with conferencing devices communicatively connected in a daisy-chain configuration using local connection ports included on each device are provided. One method involving a first conferencing device comprises receiving auxiliary mixed microphone signal(s) from at least one other conferencing device via at least one local connection port, each auxiliary signal comprising a mix of microphone signals captured by the at least one other conferencing device; determining a gain adjustment value for each auxiliary mixed microphone signal based on a daisy-chain position of the at least one other conferencing device relative to the position of the first conferencing device; adjusting a gain value for each auxiliary mixed microphone signal based on the corresponding gain adjustment value; generating a loudspeaker output signal from the gain-adjusted auxiliary mixed microphone signal(s); and providing the loudspeaker signal to the loudspeaker of the first conferencing device.

A method of audio processing includes calculating an estimated echo residual power based on a long-term amount of echo cancellation as indicated by an estimated power gain profile, where the estimated power gain profile is based on a reference signal and an echo residual signal. Echo suppression of the echo residual may then be performed based on the estimated echo residual power.

An In-Car Communication (ICC) system supports the communication paths within a car by receiving the speech signals of a speaking passenger and playing it back for one or more listening passengers. Signal processing tasks are split into a microphone related part and into a loudspeaker related part. A sound processing system suitable for use in a vehicle having multiple acoustic zones includes a plurality of microphone In-Car Communication (Mic-ICC) instances coupled and a plurality of loudspeaker In-Car Communication (Ls-ICC) instances. The system further includes a dynamic audio routing matrix with a controller and coupled to the Mic-ICC instances, a mixer coupled to the plurality of Mic-ICC instances and a distributor coupled to the Ls-ICC instances.

There is provided an audio device including a control section configured to cause an audio signal to be output, the audio signal including a sound signal obtained through playback of content and a sound signal received from a communication partner device, and a sound processing section configured to generate an elimination signal obtained by eliminating a given sound signal from a microphone detection signal, which is the audio signal that is propagated and detected by a microphone. The control section causes the communication partner device to transmit the elimination signal.

A speech signal processing method is performed at a terminal device, including: obtaining a recorded signal and a to-be-output speech signal, the recorded signal including a noise signal and an echo signal; calculating a loop transfer function according to the recorded signal and the speech signal; calculating a power spectrum of the echo signal and a power spectrum of the noise signal according to the recorded signal, the speech signal, and the loop transfer function; calculating a frequency weighted coefficient according to the two power spectra of the echo signal and the noise signal; adjusting a frequency amplitude of the speech signal based on the frequency weighted coefficient; and outputting the adjusted speech signal to a speaker electrically coupled to the terminal device. As such, the frequency amplitude of the speech signal is automatically adjusted according to the relative frequency distribution of a noise signal and the speech signal.

In an example, an audio system, which may comprise an integrated circuit, comprises an amplifier and a combiner. The amplifier is configured to output a first amplified audio signal to a speaker. The combiner is configured to: receive the first amplified audio signal from the amplifier, receive a second audio signal, combine the first amplified audio signal and second audio signal into a combined signal, and output the combined signal.