Abstract: The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal.

Abstract: Loud sounds with fast rise times, like gunfire and explosions, can cause noise-induced hearing loss (NIHL). Unfortunately, current models do not adequately explain how impulsive sounds cause NIHL, which makes it difficult to predict and prevent NIHL on battlefields and other hostile or rugged environments. Fortunately, the impulsive sounds experienced by soldiers and others working in rugged environments can be recorded using a compact, portable system that acquires, digitizes, and stores high-bandwidth audio data. An example of this system can be mounted on a helmet or other article and used to record hours of audio data at a bandwidth of 20 kHz or higher, which is broad enough to capture sounds with rise times less than 50 ms. An analog-to-digital converter (ADC) digitizes these broadband audio signals at rate of 40 kHz or higher to preserve the impulse information. A processor transfers the digitized samples from a buffer to a memory card for later retrieval using an interrupt-driven processing technique.

Abstract: In an example implementation, a method may involve sending a message that arranges a media playback system into a pre-defined state, wherein a first playback device is in a first zone having a first zone name, a second playback device is in a second zone having a second zone name, and a queue has a particular set of media items. The method may also involve detecting a trigger indicating a request to reset the media playback system. The method may further involve causing the media playback system to revert to the pre-defined state, wherein reverting may include assigning the first zone name to the first zone and joining the first playback device to the first zone, assigning the second zone name to the second zone and joining the second playback device to the second zone, and configuring the queue with the particular set of media items.

Abstract: A user-configurable modular audio control surface comprises master modules for controlling global surface properties and channel modules for controlling one or more individual audio channels. The modules are disposed in a two-dimensional spatial arrangement such that any module can occupy a location within the control surface not occupied by another module. The modules are connected to each other and to external platforms hosting media applications and plug-ins via a network. Control surface users can interact with external applications via remote graphical user interfaces displayed on modules within the surface, and can automate multiple external applications using an automation system built into the surface. Automation line graphs and metadata for both internal and external applications are displayed over the corresponding waveform displays that can include audio ahead of a current playback location.

Abstract: Systems, devices, and methods are disclosed for establishing a communication channel between a computer device associated with a hearing-care professional and various devices associated with a communication service provider for providing services to hearing-impaired users. The computer device associated with the hearing-care professional may be configured to generate at least one of a professional certification form and audiogram data through its associated software tools to the various devices associated with the communication service provider.

Abstract: 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.

Abstract: The present disclosure relates in general to media streaming, such as music streaming. In particular, this disclosure presents various embodiments of methods, portable electronic devices (200), computer servers (300) and computer program that allow for identifying an audio source (e.g., a music source) that is currently outputting audio (e.g., playing music). In an example scenario, this may allow a user to identify a social gathering such as a party where music is being played.

Abstract: A hearing device comprises a receiver, an input buffer and a sample processor, the receiver being adapted to receive samples of a digital audio signal and feed received samples as a digital input signal to the input buffer, the sample processor being adapted to process the buffered samples to provide samples of a digital output signal such that the digital output signal is a sample-rate converted representation of the digital input signal with a predetermined target sample rate. The hearing device further comprises a latency controller adapted to estimate the quality of reception of the digital audio signal and to control the processing of the buffered samples in dependence on the estimated quality of reception.

Abstract: According to embodiments described in the specification, a mobile electronic device and method for controlling a media player device to play back media are provided. The mobile device includes: a network interface; a memory storing first audio data and second audio data; a microphone; and a processor interconnected with the network interface, the memory and the microphone, the processor operating to: transmit, to a media player device via the network interface, a request to initiate a connection between the mobile device and the media player device; send the first audio data over the connection to the media player device for playback by the media player device; detect the playback of the first audio data via the microphone; and in response to detecting playback of the first audio data, send the second audio data over the connection to the player device for playback by the media player device.

Abstract: The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal.

Abstract: An audio identification system accounts for an audio signal's complexity when generating a test audio fingerprint for identification of the audio signal. In particular, the audio identification system determines a complexity of an audio signal to be fingerprinted. For example, the audio signal's complexity may be determined by performance of an autocorrelation on the audio signal. Based on the determined complexity, the audio identification system determines a length of a sample of the audio signal used to generate a test audio fingerprint. A sample having the length is then obtained and used to generate a test audio fingerprint for the audio signal. The test audio fingerprint may be compared to a set of reference audio fingerprints to identify the audio signal.

Abstract: Method and devices for testing a headphone with increased sensation are provided. The headphone can filter and amplify low frequency audio signals, which are then sent to a haptic device in the headphone. The haptic device can cause bass sensations at the top of the skull and at both ear cups. The testing system can evaluate the haptic and acoustic sensations produced by the headphone to evaluate if they have been properly assembled and calibrate the headphones if necessary.

Abstract: An example method includes receiving an indication that a user was at a location at a particular time; identifying at least one portion of media content playing at the location at the particular time; and transmitting data that indicates the identified at least one portion of media content to a computing device associated with the user. A second example method includes transmitting an indication that a user was at a location at a particular time; receiving data identifying at least one portion of media content played by a second playback device at the location at the particular time when the user was at the location; and causing, by the control device, a graphical display to display the data identifying the at least one portion of media content played at the location at the particular time when the user was at the location.

Abstract: This disclosure relates to audio identification using ordinal transformations. A media matching component receives a sample audio file. The sample audio file can include, for example, a cover song. The media matching component includes a vector component that computes a set of vectors using auditory feature values included in the sample audio file. A hashing component employs a hash function to generate a fingerprint, including a set of sub-fingerprints, for the sample audio file using the set of vectors. The fingerprint is invariant to variations including but not limited to variations in key, instrumentation, encoding formats, performers, performance conditions, arrangement, and/or recording and processing variations. An identification component determines if any reference audio files are similar to the sample audio file using the fingerprint and/or sub-fingerprints, and identifies any similar reference audio files.

Abstract: Users may have a set duration during which they may consume content, or they may have a variable duration during which they may consume content. A content pacing service is disclosed so that a portion of an item of content may be conveyed in the set duration during which a user is to consume the content. In one embodiment, the rate at which the content is conveyed is increased or decreased so that a reference point in the item of content (e.g., the end of a chapter in an audiobook) is reached approximately when the duration of the user's content consumption ends.

Abstract: A method and an audio conference device for carrying out an audio conference are disclosed, whereby classification information associated with a respective audio date flow is recorded for supplied audio data flows. According to a result of an evaluation of the classification information, the audio data flows are associated with at least three groups which are homogeneous with regard to the results. The individual audio data flows are processed uniformly in each group in terms of the signals thereof, and said audio data flows processed in this way are superimposed in order to form audio conference data flows to be transmitted to the communication terminals.

Abstract: A method and an audio conference device for carrying out an audio conference are disclosed, whereby classification information associated with a respective audio date flow is recorded for supplied audio data flows. According to a result of an evaluation of the classification information, the audio data flows are associated with at least three groups which are homogeneous with regard to the results. The individual audio data flows are processed uniformly in each group in terms of the signals thereof, and said audio data flows processed in this way are superimposed in order to form audio conference data flows to be transmitted to the communication terminals.

Abstract: A method of decoding an audio signal, and which includes extracting spectral data and a loss signal compensation parameter from an audio signal bitstream; detecting a loss signal based on the spectral data; generating first compensation data corresponding to the loss signal using a random signal based on the loss signal compensation parameter; generating a scale factor by adding a scale factor difference value to a scale factor reference value if the scale factor corresponds to a band quantized to zero, and the scale factor reference value is included in the loss signal compensation parameter; and generating second compensation data by applying the scale factor to the first compensation data.

Abstract: A data transmitting apparatus includes a packet generator which generates a plurality of packets regarding multi-channel audio sample data and transmitter which transmits the plurality of generated packets to a data receiving apparatus. Each of the plurality of packets includes an identifier field to distinguish a position or an order of the packets.

Abstract: A reproduction system has a content reproduction section 220 that reproduces a content having a cycle; a special reproduction operation section 230 that performs special reproduction on the content by which the content is reproduced at a reproduction position different from a normal reproduction return position at a time of normal reproduction reproduced normally; a special reproduction process section 235 that makes the content reproduction section 220 perform the special reproduction based on an operation of the special reproduction operation section 230; and a return process section 250 that matches the normal reproduction return position of the content with a position corresponding to cycle timing with which the normal reproduction continues without the special reproduction when the normal reproduction returns after the special reproduction is cancelled in case that the special reproduction is performed.