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.

A speech signal leveling system and method include generating an output signal by applying a frequency-dependent or frequency-independent controllable gain to an input signal, the gain being dependent on a gain control signal, and generating at least one speech detection signal indicative of voice components contained in the input signal. The system and method further include generating the gain control signal based on the input signal and the at least one speech detection signal, controlling the controllable-gain block to amplify or attenuate the input signal to have a predetermined mean or maximum or absolute peak signal level as long as voice components are detected in the input signal.

Disclosed are systems and methods for processing an audio signal. In particular, there is provided a method for determining dynamic gain values to be applied on a digital input signal. The digital signal may be arranged in blocks. The dynamic gain values may be used for attenuating input signal values exceeding a clipping threshold. More particularly, the method comprising, for each signal block, passing backwards over the next signal block and the current signal block to produce a preliminary gain contour from the input signal; and passing forwards over the current signal block to produce a final gain contour for the current signal block based on the preliminary gain contour, wherein the gain contours are produced by applying an instant gain ascent and a smooth gain decay to the gain contours.

Systems and methods for leveling loudness variation in an audio signal are described. Embodiments use both a perceptual leveling algorithm and a standards-based loudness measure together to minimize audio process artifacts and ensure that the measured loudness of the processed audio is close to a required measure, according to a particular standard measurement of loudness. These systems and methods can be used either offline or in real-time.

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.

Apparatuses and methods are described to identify desired audio. A first input of an apparatus is configured to receive a main signal. A second input of the apparatus is configured to receive a reference signal. A normalizer is configured to normalize a compressed main signal by a compressed reference signal to create a normalized main signal. A single channel normalized voice threshold comparator is configured to receive as an input the normalized main signal and to output a desired voice activity detection signal.

Synchronizing ancillary data to content including audio includes obtaining a representation of the content's audio and ancillary data pegged to instants in the representation of the content's audio, and aligning the representation of the content's audio to the content's audio to synchronize the ancillary data pegged to the instants in the representation of the content's audio to the content.

Example embodiments provide a process that includes one or more of receiving an audio signal at a feedback compressor circuit, multiplying the received audio signal with a power feedback signal to create a product audio signal, wherein the feedback signal comprises a low-pass filtered signal, applying a power amplifier to the product audio signal, and providing the amplified product audio signal as an output signal to a speaker.

An electronic apparatus and a method for activating a specified function are provided. The electronic apparatus includes a speaker, an audio signal processor and an application processor. The audio signal processor senses a variation of an acoustic condition of the speaker. The application processor is used for: generating a logic high or low signal in response to the sensed variation of the acoustic condition; interpreting the logic high or low signal as a control signal; and performing an instruction corresponding to the control signal.

A wireless communication device is disclosed. The wireless communication device includes a processor, a memory, a transceiver configured to receive an audio signal, a codec to decode the audio signal, a dynamic range controller and a phoneme processor. The phoneme processor is configured to extract acoustic cues from each frame of the decoded audio signal and to identify a phoneme class in the each frame. The dynamic range controller is configured to apply dynamic range compression on the each frame based on the identified phoneme class.