Abstract:
In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.
Abstract:
Dynamic loudness equalization of received audio content in a playback system, using metadata that includes instantaneous loudness values for the audio content. A playback level is derived from a user volume setting of the playback system, and is compared with a mixing level that is assigned to the audio content. Parameters are computed, that define an equalization filter that is filtering the audio content before driving a speaker with the filtered audio content, based on the instantaneous loudness values and the comparing of the playback level with the assigned mixing level. Other embodiments are also described and claimed.
Abstract:
One embodiment provides a method, including: detecting, using a processor, a presence of a known device within a predetermined range; and adjusting, using the processor, an audio output volume based on the presence of the known device. Other aspects are described and claimed.
Abstract:
A system for encoding and applying Dynamic Range Control/Compression (DRC) gain values to a piece of sound program content is described. In particular, a set of DRC gain values representing a DRC gain curve for the piece of content may be divided into frames corresponding to frames of the piece of content. A set of fields may be included with an audio signal representing the piece of content. The additional fields may represent the DRC gain values using linear or spline interpolation. The additional fields may include 1) an initial gain value for each DRC frame, 2) a set of slope values at particular points in the DRC curve, 3) a set of time delta values for each consecutive pair of slope values, and/or 4) one or more gain delta values representing changes of DRC gain values in the DRC gain curve between points of the slope values.
Abstract:
A level of an input signal is detected according to a first following rate, and a first level signal indicating the detected level is generated. A level of the input signal is detected according to a second following rate lower than the first following rate, and a second level signal indicating the detected level is generated. One of the first and second level signals is selected based on a relation (e.g., ratio) between the first and second level signals so that a gain is determined based on the selected one of the first and second level signals. The level of the first input signal is adjusted according to the determined gain. For example, if the level variation is dominant, the gain adjustment suitable for the level variation can be performed, whereas, if the stable level is dominant, the gain adjustment suitable for the stable level can be performed.
Abstract:
Embodiment disclosed herein enable detection and improvement of the quality of the audio signal using a mobile device by determining the loss in the audio signal and enhancing audio by streaming the remainder portion of audio. Embodiments disclosed herein enable an improvement in the sound quality rendered by rendering devices by emitting an test audio signal from the source device, measuring the test audio signal using microphones, detecting variation in the frequency response, loudness and timing characteristics using impulse responses and correcting for them. Embodiments disclosed herein also compensate for the noise in the acoustic space by determining the reverberation and ambient noise levels and their frequency characteristics and changing the digital filters and volumes of the source signal to compensate for the varying noise levels.
Abstract:
An example implementation involves a first playback device receiving audio content to be played back by the first playback device and a second playback device in synchrony. The audio content includes a first stereo component to be played by the first playback device and a second stereo component to be played by the second playback device. The first playback device determines a first limiting result that represents playback of the second stereo component by a second playback device by applying, to the second stereo component, a pre-determined volume-limiting function that is associated with the second playback device. The first playback device determines another volume-limiting function based on the first limiting result and applies the determined volume-limiting function to the first stereo component when playing the first stereo component in synchrony with the playback of the second stereo component by the second playback device.
Abstract:
Volume limiting systems and methods are operable to limit volume output from media presentation devices. An exemplary embodiment detects a sound using a microphone, wherein the sound corresponds to an audio output of at least one controlled media presentation device, and wherein the microphone is remotely located from the at least one controlled media presentation device; compares a level of the detected sound with a predefined maximum volume limit; generates a volume output limit command in response to the detected sound exceeding the predefined maximum volume limit; and communicates the volume output limit command to the media presentation device. The media presentation device then reduces a volume level of its audio output. In some instances, volume may be limited during user specified periods.
Abstract:
An amplifier arrangement for amplifying an audio input signal AES into an audio output signal AAS, having a conditioning apparatus for converting the audio input signal AES into a conditioned intermediate signal ZS. The conditioning apparatus includes an audio input interface for accepting the audio input signal and a digital data processing device. The amplifier arrangement also includes an amplifier apparatus for amplifying the intermediate signal ZS into the audio output signal AAS and the amplifier apparatus has at least one operating voltage BS. The amplifier arrangement also includes a limiting module for limiting the audio output signal AAS by changing a gain parameter VK.
Abstract:
Level adjusting circuit for adjusting level of input audio signal, includes: N filters, N being integer of two or more; N dynamic range compression (DRC) circuits corresponding to N filters; adder; (N−1) band pass filters corresponding to crossover frequencies of N filters; and (N−1) gain correcting units corresponding to (N−1) band pass filters. Each filter receives and passes input audio signal through its respective set band. ith (1≦i≦N) DRC circuit amplifies signal from ith filter and adjust its gain to prevent level of its output signal from exceeding threshold level. Adder adds each of output signals of DRC circuits. jth (1≦j≦N−1) band pass filter passes band including crossover frequencies of jth and (j+1)th filters from output signal of adder. jth gain correcting unit adjusts at least one of gains of jth and (j+1)th DRC circuits based on level of output signal of jth band pass filter.