The invention relates to a method for rendering ultrasonic signals audible that is characterized in that the temporal dynamic range of the ultrasonic signal is maintained. The amplitude profile of the ultrasonic signal picked up in the time domain remains unaltered. The frequency shift from the ultrasonic range to the audible range is possible up to a factor of 32 using the present invention.

A headphone with adjustable speaker drivers and a microphone can be used to determine and adjust sound pressure levels. The speaker drivers can be adjusted manually or wirelessly via a mobile device with a wireless connection to the headphone. Processing of audio and microphone data via the headphone can also be used to help determine and adjust the sound pressure levels.

Methods, systems, and computer program products that infer and correct automatic gain compensation (AGC) values over time are described. A device emits a series of inaudible reference audio signals during recording. The reference audio signals have a constant amplitude. A microphone of the device records the reference audio signals while recording audio content. The device may apply AGC during the recording. An AGC inference module receives the recorded signals and extracts a representation of the reference audio signals, which have been subject to the AGC. The AGC inference module determines variations in levels of the extracted representation of the reference audio signals over time. Based on the variations, the AGC inference module infers the AGC applied to the recording over time. The AGC inference module can then provide the AGC for reading, or undo effects of the AGC applied to the audio content.

An example method includes receiving data indicating a configuration of one or more playback devices. The one or more playback devices may include one or more transducers. The method further includes, based on the received data, associating each of one or more audio streams respectively with at least one transducer of the one or more transducers. The method further includes generating the one or more audio streams and sending at least one of the generated one or more audio streams to each of the one or more playback devices. An example non-transitory computer readable medium and an example computing device related to the example method are also disclosed herein.

A wireless audio device is provided. The wireless audio device includes an audio receiving circuit, an audio output circuit, an acceleration sensor, a communication circuit, a processor, and a memory. The memory may store instructions that, when executed by the processor, cause the wireless audio device to detect an utterance of a user of the wireless audio device by using the acceleration sensor, enter a dialog mode in which at least some of ambient sounds received by the audio receiving circuit are output through the audio output circuit, in response to detecting the utterance of the user, and end the dialog mode if no voice is detected for a specified time or longer by using the audio receiving circuit in the dialog mode.

In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

An audio encoder device includes an audio encoder configured for producing an encoded audio bitstream from an audio signal having consecutive audio frames; a dynamic range control encoder configured for producing an encoded dynamic range control bitstream from .[.an.]. .Iadd.a .Iaddend.dynamic range control sequence corresponding to the audio signal and having consecutive dynamic range control frames, wherein each dynamic range control frame of the dynamic range control frames has one or more nodes, wherein each node of the one or more nodes has gain information for the audio signal and time information indicating to which point in time the gain information corresponds.

A method for performing loudness normalization by an audio signal processing device includes: receiving an audio signal; receiving information relating to integrated loudness of the audio signal; receiving information relating to target loudness of the audio signal; obtaining a corrected integrated loudness by correcting the integrated loudness based on one or more audio processing; and obtaining a normalized audio signal by normalizing the audio signal based on the corrected integrated loudness and the target loudness.

Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

An audio processing system includes a server complex in communication with a network. The server complex receives a digital audio file and one or more analog domain control settings from a client device across the network. A digital-to-analog converter converts the digital audio file to an analog signal. One or more analog signal processors apply at least one analog modification to the analog signal in accordance with the one or more analog domain control settings. An analog-to-digital converter converts the modified analog signal to a modified digital audio file. The server complex can then deliver the modified digital audio file to the client device across the network.