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.

This application relates to Class D amplifier circuits. A modulator controls a Class D output stage based on a modulator input signal (Dm) to generate an output signal (Vout) which is representative of an input signal (Din). An error block, which may comprise an ADC, generates an error signal (ε) from the output signal and the input signal. In various embodiments the extent to which the error signal (ε) contributes to the modulator input signal (Dm) is variable based on an indication of the amplitude of the input signal (Din). The error signal may be received at a first input of a signal selector block. The input signal may be received at a second input of the signal selector block. The signal selector block may be operable in first and second modes of operation, wherein in the first mode the modulator input signal is based at least in part on the error signal; and in the second mode the modulator input signal is based on the digital input signal and is independent of the error signal. The error signal can be used to reduce distortion at high signal levels but is not used at low signal levels and so the noise floor at low signal levels does not depend on the component of the error block.

Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to the substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

The present invention provides methods and systems for digital processing of an input audio signal. Specifically, the present invention includes a high pass filter configured to filter the input audio signal to create a high pass signal. A first filter module then filters the high pass signal to create a first filtered signal. A first compressor modulates the first filtered signal to create a modulated signal. A second filter module then filters the modulated signal to create a second filtered signal. The second filtered signal is processed by a first processing module. A band splitter splits the processed signal into low band, mid band, and high band signals. The low band and high band signals are modulated by respective compressors. A second processing module further processes the modulated low band, mid band, and modulated high band signals to create an output signal.

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 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 technique, suitable for real-time processing, is disclosed for pitch tracking by detection of glottal excitation epochs in speech signal. It uses Hilbert envelope to enhance saliency of the glottal excitation epochs and to reduce the ripples due to the vocal tract filter. The processing comprises the steps of dynamic range compression, calculation of the Hilbert envelope, and epoch marking. The Hilbert envelope is calculated using the output of a FIR filter based Hilbert transformer and the delay-compensated signal. The epoch marking uses a dynamic peak detector with fast rise and slow fall and nonlinear smoothing to further enhance the saliency of the epochs, followed by a differentiator or a Teager energy operator, and amplitude-duration thresholding. The technique is meant for use in speech codecs, voice conversion, speech and speaker recognition, diagnosis of voice disorders, speech training aids, and other applications involving pitch estimation.

The present technology relates to a signal processing device, a signal processing method, and a program for enabling reproduction of high-quality sounds with a low process load. The signal processing device includes a demultiplexing section that extracts encoded audio signals and overamplitude flags, which have been generated for a plurality of respective panel loudspeakers and each indicate whether overamplitude will occur in the corresponding panel loudspeaker, by demultiplexing encoded data, a decoding section that decodes the encoded audio signals, and an adjustment section that adjusts audio signals to be supplied to the plurality of panel loudspeakers on the basis of the overamplitude flags and audio signals resulting from the decoding. The present technology is applicable to an encoding device and a decoding device.

A method for normalizing platform-adaptive audio includes encoding input video content and generating video stream data as original data to store the video stream data in storage; generating loudness metadata for audio data of the video content and storing the loudness metadata in the storage; receiving a request for the video content from a client; searching the storage for video stream data of the video content corresponding to the request, the loudness metadata, and a device profile corresponding to device information included in the request; and transmitting, to the client, a response including the video stream data, the loudness metadata, and the device profile that are found in the storage.

Disclosed is a method of operating an audio signal processing apparatus playing content including an audio signal. The method includes receiving the audio signal, receiving metadata including information related to a loudness of the audio signal, the metadata including loudness distribution information indicating, for each of a plurality of steps separated according to a loudness magnitude, a ratio between an amount of the audio signal corresponding to each of the plurality of steps of the audio signal and a total amount of the audio signal, and adjusting the loudness of the audio signal based on the metadata.

Example embodiments provide a process that includes one or more of receiving an audio signal from a feedback path of a feedback compressor circuit, determining whether an auxiliary attenuation value applied to the feedback compressor circuit has changed since a last audio signal was received, responsive to determining the auxiliary value has changed, determining a current operational state value of the LPF needs to be modified based on the changed auxiliary attenuation value, modifying the operational state value of the LPF, and applying the audio signal to the modified LPF.