Abstract: A digital mixer includes a MO switch and a MO fader in a signal transmission route from each input channel to a bus of a MIX_MINUS type, a signal of the input channel is supplied to the MIX bus without control on its level when the MO switch is OFF, while being supplied to the MIX bus after being controlled on its level by the MOL fader when the MO switch is ON. When the MO switch is changed from OFF to ON, a level adjustment amount by the MOL fader is set to a value so that a signal passing therethrough comes to have zero level.

Abstract: A user device that plays back audio books for a user can include a dynamic pause that provides a user with greater flexibility in when to pause playback of an audio book. Dynamic pause includes initiating playback of an audio book using a user device; receiving a pause request as input to the user device, the pause request received at an input time index during playback of the audio book; retrieving a subset of candidate pause points, each candidate pause point comprising a time index within the audio book that corresponds to a break point located within an eBook corresponding to the audio book; selecting one of the candidate pause points from the subset, the time index of the selected candidate pause point determining a pause time index when playback is to be paused; and pausing the playback at the pause time index.

Abstract: Example methods and systems for modifying the playback of content using pre-processed profile information are described. In some example embodiments, the methods and systems may access a stream of content to be delivered to a playback device, identify a piece of content within the stream of content to be delivered to the playback device, determine a profile for the identified piece of content, and deliver the determined profile to the playback device. In some example embodiments, the methods and systems may receive a stream of content at a playback device, access profile information associated with the stream of content, and modify playback of the stream of content based on the accessed profile information.

Abstract: In a hardware audio CODEC which processes audio signals from a plurality of inputs, voltage and/or power levels of the input audio signals may be adjusted such that the digitally adjusted levels are approximately equal for each of the plurality of inputs. The digital adjustment may comprise, for each audio sample of one of the input audio signals, adding the audio sample to one or more right shifted versions of the audio sample and selecting a portion of a summed audio signal resulting from the addition. The portion of the summed audio that is selected may be determined based on the type of audio content being processed. The one or more right shifted versions of the audio sample may be selected via one or more switching elements which may be controlled via a digital control word which may be dynamically generated.

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: Sound alignment techniques that employ timing information are described. In one or more implementations, features and timing information of sound data generated from a first sound signal are identified and used to identify features of sound data generated from a second sound signal. The identified features may then be utilized to align portions of the sound data from the first and second sound signals to each other.

Abstract: In general, techniques of controlling a plurality of multimedia players in groups are disclosed. According to one aspect of the present invention, a mechanism is provided to allow a user to group some of the players according to a theme or scene, where each of the players is located in a zone. When the scene is activated, the players in the scene react in a synchronized manner. For example, the players in the scene are all caused to play a multimedia source or music in a playlist, wherein the multimedia source may be located anywhere on a network.

Abstract: A process includes receiving, from a user identity, instructions for output characteristics including one or more of audio characteristics for rendering or capturing audio data or visual characteristics for rendering or capturing visual data. The process also includes determining, in response to the received instructions, output controls which effect the one or more of audio characteristics or visual characteristics, and associating the output controls with an online identity or resource. The process further includes storing the associated output controls and detecting an interaction with the online identity or resource. Moreover, the process includes accessing, in response to the detection of the interaction, the stored output controls, and enabling an effect, based on the output controls, of one or more of the audio characteristics or the visual characteristics with respect to interaction with the online identity or resource.

Abstract: Methods, systems and computer readable media for altering an audio output are provided. In some embodiments, the system may change the original frequency content of an audio data file to a second frequency content so that a recorded audio track will sound as if a different person had recorded it when it is played back. In other embodiments, the system may receive an audio data file and a voice signature, and it may apply the voice signature to the audio data file to alter the audio output of the audio data file. In that instance, the audio data file may be a textual representation of a recorded audio data file.

Abstract: As viewed from the front of an operation panel, adjustment sections for individual input channels are positioned on the operation panel close to a human operator, and a display, a control section for setting an equalizer process, compressor process etc. for one selected channel and a signal delivery control section for setting signal delivery to mixing buses for the selected channel are positioned in a horizontal row remotely from the human operator. Operators in the delivery control section are varied in function in accordance with a selected operation mode. For example, in a mix send mode, a plurality of rotary operators are set to function as send level adjustment operators for the respective input channels so as to adjust send levels to a given mixing bus, while, in a mix master mode, the rotary operators are set to function as output gain adjustment operators for the respective mixing buses.

Abstract: An output channel processing an audio signal after mixing in a mixing bus in a digital mixer is configured such that output points PreHPF and PostON are provided at locations before and after a signal processing module group composed of signal processing modules from a high-pass filter to an ON/OFF control module, an output selecting switch selects one of the output points to supply the audio signal at the selected output point to a direct-out output module which is provided corresponding to the output channel and includes an ON/OFF control module and a level adjustment module so that the audio signal can be outputted via the direct-out output module.

Abstract: Methods, systems, and computer program products are provided for supplementing audio recorded in a media file. Embodiments include receiving a media file; identifying the subject matter of audio portion of the media file; identifying supplemental content for supplementing the subject matter recorded in the audio portion of the media file; and inserting in the media file markup for rendering the supplemental content.

Abstract: A method of processing an audio signal, and which includes a receiving object information including object level information indicating a level of objects, object gain information and object correlation information; receiving preset presence information and preset number information; receiving preset information based on the preset presence information and the preset number information by an audio decoding apparatus, the preset information including preset metadata representing an attribute mode of the preset information, preset rendering data and preset type information indicating whether the preset rendering data is in a matrix type or not obtaining a preset matrix from the preset rendering data by the audio decoding apparatus, and generating a processed audio signal by controlling a gain or panning of the at least one object included in the audio signal based on the downmix processing information.

Abstract: Techniques are disclosed for acoustic data reception. Playback of modified audio content is acoustically detected by a first receiving entity and a second receiving entity operatively connected to the first receiving entity via a network. The modified audio content is generated by encoding specified data into specified audio content such that the modified audio content satisfies a predefined signal constraint characterizing imperceptibility of any differences between playback of the specified audio content and playback of the modified audio content. The specified data is collaboratively decoded based on the acoustically detected playback of the modified audio content and by the first receiving entity and the second receiving entity using the network and according to a predefined diversity scheme.

Abstract: A media signal is accessed, which has been generated with one or more first processing operations. The media signal includes one or more sets of artifacts, which respectively result from the one or more processing operations. One or more features are extracted from the accessed media signal. The extracted features each respectively correspond to the one or more artifact sets. Based on the extracted features, a conditional probability score and/or a heuristically based score is computed, which relates to the one or more first processing operations.

Abstract: An audio signal distortion detection device for simplifying the detection of harmonic distortion to aid in setting an amplifier's gain control to match an audio source unit's output voltage. The audio signal distortion detection device generally includes a housing having a plurality of status indicators thereon for indicating the status of an audio signal being fed into the housing's audio input. By utilizing a scaling circuit and processing unit, the present invention may be configured to detect four audio frequencies and compare DC levels of the frequencies to each other to determine if harmonic distortion is present. A distortion illuminator will illuminate when such distortion is detected and thus significantly reduce the effort required to setting an amplifier's gain control.

Abstract: A web-based method of combining time coded audio files, having the following steps: (a) a local operator using a web-based interface displayed on a local computer system to access a library of audio files stored on a remote system; (b) the local operator using the web-based interface to select a plurality of the audio files from the library; (c) the local operator using the web-based interface to request the remote system to combine the selected audio files into a single combined audio segment comprising the selected audio files commencing play at audio cue markers; (d) the remote system generating the single combined audio segment by combining the plurality of selected audio files; and (e) the local operator using the web-based interface to download or stream the single combined audio segment.

Abstract: A cable for transmitting a transmissible element, such as power or data, from a source to a device, the cable includes at least a first plug configured to mate with at least one of the source and the device, and an audio player for playing back audio recording.

Abstract: A method, an apparatus, a system, and instructions stored in a non-transitory computer-readable medium to instruct a processing system to carry out the method. The method includes applying corrective filters directly in a portable media device to correct, e.g., equalize for the overall system comprising the portable media device and the playback system to which it is attached. Also a method of determining the corrective filters by playing back one or more calibration signals on the playback system while recording the resulting sound field on the portable media device.

Abstract: A method and apparatus to encode audio with spatial information in a manner that does not depend on the exhibition setup, and to decode and play out optimally for any given exhibition setup, maximizing the sweet-spot area, and including setups with loudspeakers at different heights, and headphones. The part of the audio that requires very precise localization is encoded into a set of mono tracks with associated directional parameters, whereas the remaining audio is encoded into a set of Ambisonics tracks of a chosen order and mixture. Upon specification of a given exhibition system, the exhibition-independent format is decoded adapting to the specified system, by using different decoding methods for each assigned group.