303results about "Volume compression/expansion in digital/coded amplifiers" patented technology

An audio normalization gain value is applied to an audio signal to produce a normalized signal. The normalized signal is processed to compute dynamic range control (DRC) gain values in accordance with a selected one of several pre-defined DRC characteristics. The audio signal is encoded, and the DRC gain values are provided as metadata associated with the encoded audio signal. Several other embodiments are also described and claimed.

A method and system for selectively and variably attenuating audio data are disclosed. A high-volume control value as selected by a user is first received, and this high-volume control value defines a volume output level for high amplitude audio samples. An attenuation factor is then determined by utilizing the high-volume control value. Each sample from an incoming audio data stream is conditionally attenuated with the attenuation factor such that high amplitude audio data get compressed while low amplitude audio data remain unaffected. Finally, the attenuated samples are sent to an output.

Output control system and method includes calculating gain for an input signal, detecting a predetermined condition, and modifying the gain calculation in accordance with one or more kneepoints to provide a varying output signal. A companding ratio is then established in accordance with the kneepoints. System gain is then adjusted, and the companding ratio may be further adjusted by adjusting one of the kneepoints relative to another.

A mobile audio device (for example, a cellular telephone, personal digital audio player, or MP3 player) performs Audio Dynamic Range Control (ADRC) and Automatic Volume Control (AVC) to increase the volume of sound emitted from a speaker of the mobile audio device so that faint passages of the audio will be more audible. This amplification of faint passages occurs without overly amplifying other louder passages, and without substantial distortion due to clipping. Multi-Microphone Active Noise Cancellation (MMANC) functionality is, for example, used to remove background noise from audio information picked up on microphones of the mobile audio device. The noise-canceled audio may then be communicated from the device. The MMANC functionality generates a noise reference signal as an intermediate signal. The intermediate signal is conditioned and then used as a reference by the AVC process. The gain applied during the AVC process is a function of the noise reference signal.

A signal processing method including the steps of: receiving a first audio signal; detecting the presence of one or more shrieks within an audible frequency range of said audio signal; creating one or more filters to selectively attenuate the respective one or more shrieks within the audible frequency range; filtering the audio signal using the one or more filters; and transmitting the filtered audio signal to an audio telephone device.

An example of a method of correcting an audio level of a stored program asset comprises retrieving a stored program asset having audio encoded at a first loudness setting. Dialog of the audio of the asset is identified, a loudness of the dialog is determined and the determined loudness is compared to the first loudness setting. The asset is re-encoded at a second loudness setting corresponding to the determined loudness, if the first loudness setting and the second loudness are different by more than a predetermined amount. The determined loudness is preferably a DIALNORM of the dialog. The asset may be stored with the re-encoded loudness setting. The method may be applied to programs as they are being received from a source, as well. Aspects of the method may also be applied to programs to be provided by a source. Systems are also disclosed.

A method and system are provided for equalizing the loudness of an audio source. Initially, the perceptual loudness level of an audio signal is measured from one or more audio sources. Next, the loudness level of the audio signal is adjusted using the perceptual loudness level. Thereafter, the audio signal corresponding to the music selections is reproduced such that the perceived loudness to a listener is the same entirely throughout a music track corresponding to the music selections.

The overall loudness of an audio track is calculated by combining a number of weighted loudness measures for segments of the audio track, where the weight applied to each individual loudness measure is a function of the loudness measure. By comparing the original overall loudness measure to a desired overall loudness measure, a gain can be determined that will adjust the loudness level to the desired value. Also disclosed is a dynamic compression method that analyzes the dynamic characteristics of an audio track and determines appropriate compressor parameters. Additionally, the loudness of a post-compressor audio track can be estimated for any given compressor parameters, thus permitting post-compression loudness matching to be done even if the compression is performed in real-time.

In one disclosed aspect, dynamic gain modifications are applied to an audio signal at least partly in response to auditory events and / or the degree of change in signal characteristics associated with said auditory event boundaries. In another aspect, an audio signal is divided into auditory events by comparing the difference in specific loudness between successive time blocks of the audio signal.

The invention relates to a method and an arrangement intended for radio communication systems and effective in digitalizing and subsequently processing numerically arbitrary radio signals. The signals are represented by composite (complex) vectors which have been subjected to disturbances in the system, such that information in the signals has been lost. This information is restored in its entirety when practising the present invention. For the purpose of solving this problem, the inventive digitalizing arrangement includes a multistage logarithmic amplifier chain (A) in which each stage is connected to a separate detector (D), the output signals of which are added in an adder. The adder output signals are then transmitted to a first A / D-converter (AD1) for digitalizing and converting the amplitude components of the signal. At the same time, the undetected signal from the saturated output stage in the amplifier chain is transmitted to a second A / D-converter for digitalizing and converting the phase components of the signal. The digital values obtained on the outputs of the AD-converters (AD1, AD2) are applied to different inputs of a digital signal processor (MP) for numerical processing of the pairwise received digital values in a manner such as to restore the complete vector characteristic of the signal.

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.

The present invention provides for methods and systems for digitally processing an audio signal. Specifically, the present invention provides for a speaker system that is configured to digitally process an audio signal in a manner such that studio-quality sound that can be reproduced.

The invention provides one sound producing system including a speaker, a driving circuit and an audio controller. The speaker is used for producing an audible sound. The driving circuit is connected to the speaker. The audio controller is configured to receive an input audio signal and pre-amplify the input audio signal with an amplify gain to obtain a pre-amplified audio signal; operate a multiband dynamic range control on the pre-amplified audio signal to obtain an output audio signal; convert the output audio signal to a driving voltage; provide the driving voltage to the speaker through the driving circuit; detect over the driving circuit to obtain at least one excursion parameter to determine an estimated excursion of the speaker in response to the driving voltage; and adjust the amplify gain according to the estimated excursion.

A mobile audio device (for example, a cellular telephone, personal digital audio player, or MP3 player) performs Audio Dynamic Range Control (ADRC) and Automatic Volume Control (AVC) to increase the volume of sound emitted from a speaker of the mobile audio device so that faint passages of the audio will be more audible. This amplification of faint passages occurs without overly amplifying other louder passages, and without substantial distortion due to clipping. Multi-Microphone Active Noise Cancellation (MMANC) functionality is, for example, used to remove background noise from audio information picked up on microphones of the mobile audio device. The noise-canceled audio may then be communicated from the device. The MMANC functionality generates a noise reference signal as an intermediate signal. The intermediate signal is conditioned and then used as a reference by the AVC process. The gain applied during the AVC process is a function of the noise reference signal.

An apparatus and method of automatically compensating audio volume in response to a channel change. The method includes generating a gain value to adjust a level of an input audio signal by comparing an average level of the input audio signal with a predetermined input / output level curve, adjusting an audio volume of the input audio signal by applying the generated gain value to the level of the input audio signal, and adjusting a dynamic range of the audio signal having the adjusted audio volume.

A frequency-warped processing system using either sample-by-sample or block processing is provided. Such a system can be used, for example, in a hearing aid to increase the dynamic-range contrast in the speech spectrum, thus improving ease of listening and possibly speech intelligibility. The processing system is comprised of a cascade of all-pass filters that provide the frequency warping. The power spectrum is computed from the warped sequence and then compression gains are computed from the warped power spectrum for the auditory analysis bands. Spectral enhancement gains are also computed in the warped sequence allowing a net compression-plus-enhancement gain function to be produced. The speech segment is convolved with the enhancement filter in the warped time-domain to give the processed output signal. Processing artifacts are reduced since the frequency-warped system has no temporal aliasing.

Sound pressure level limiter with anti-startle feature for audio systems are disclosed. The anti-startle feature may be implemented with a delta incident detector for detecting delta acoustic incidents that exceed a predetermined acoustic startle boundary, a delta limiter for determining an anti-startle gain, and an amplifier to apply the anti-startle gain to the input signal. The delta incident detector may detect delta incidents based on an estimated true SPL delivered by a transducer to a predetermined datum point. The estimated true SPL may be determined by based on a measured receiving frequency response of the transducer. An SPL limiter may also determine an SPL gain in response to detecting an SPL acoustic incident that exceeds a predetermined SPL threshold, and an amplifier may apply the SPL gain to the input signal to reduce it below the threshold.

A circuit for preventing clipping in an Automatic Level Control (ALC) or Limiter, where the amplitude of the signal above the clipping point is estimated, then the signal level is automatically reduced over a defined period substantially equal to the feedforward delay in the ALC / Limiter. By adaptively controlling, based on the excess amplitude and the delay time available, an attack rate used in the ALC / Limiter to reduce the gain applied to an input signal, it can be ensured that the output amplitude is brought within the clipping level sufficiently quickly to prevent audible clipping.

Large signal peaks (for example, peaks 105 and 110) can dramatically increase the cost of a communications device by requiring high grade power amplifiers. Peak reduction can reduce the large peaks at a cost of increasing spurious emissions. A circuit (for example, circuit 500) combining a scaling window peak reducing unit (for example, unit 507) and a filter (for example, filter 509) can effectively reduce the signal peaks without significantly increasing spurious emissions.

A system for controlling dynamic range of an audio signal comprises an automatic gain control element that receives an input signal having a varying level and outputs a control signal that varies based on the varying level of the input signal and a modified input signal having a dynamic range different than a dynamic range of the input signal. The system also comprises an inverter that inverts the control signal or a block-based control signal corresponding to the control signal in block format. The system also comprises a variable gain element that receives the modified input signal and at least some of the inverted control signal or block-based control signal. The variable gain element also outputs a remainder signal corresponding to the modified input signal as unmodified based on the at least some of the inverted control signal or block-based control signal.

A digital audio processor comprises at least one digital audio sample mapping module and produces a sequence of output digital audio sample values by applying a quasi time-invariant transfer function to each input digital audio sample value, the transfer function being arranged so that a ratio of root mean square signal level to peak signal magnitude is lower for the input digital audio signal than for a corresponding portion of the output digital audio signal.

The invention describes a method of automatically adjusting the loudness of an audio signal, which method comprises calculating loudness measures for samples of the input audio signal (1), identifying a number of distinct loudness lines (L1, L2, L3, L4) over 5 time among the loudness measures and altering the samples of the input audio signal (1) according to the identified loudness lines (L1, L2, L3, L4) to give an output audio signal (5) with adjusted loudness.

Systems and methods for reducing the noise level in a multi-channel digital audio system by staggering the timing of the pulse-width modulation in the different channels and thereby reducing the magnitude and increasing the frequency characteristics of the generated switching noise. One embodiment comprises a multi-channel digital PWM amplifier in which the timing signals used by each channel's modulator are staggered to evenly space the switching edges of the generated PWM signals. An additional, complementary delay is implemented in each of the channels to equalize the total delay for each channel so that the outputs of the channels are synchronized. The different channels may be implemented on different chips, in which case the chips may be synchronized prior to staggering the signals processed in each of the channels.

The present invention relates to a signal processing apparatus and method, a program, and a data recording medium configured such that the playback level of an audio signal can be easily and effectively enhanced without requiring prior analysis.An analyzer 21 generates mapping control information in the form of the root mean square of samples in a given segment of a supplied audio signal. A mapping processor 22 takes a nonlinear function determined by the mapping control information taken as a mapping function, and conducts amplitude conversion on a supplied audio signal using the mapping function. In this way, by conducting amplitude conversion of an audio signal using a nonlinear function that changes according to the characteristics in respective segments of an audio signal, the playback level of an audio signal can be easily and effectively enhanced without requiring prior analysis. The present invention may be applied to portable playback apparatus.

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.

A method for audio downmix apparatus with dynamic-range control includes steps as follows. The procedure first calculates the power of a surround audio channel and a central audio channel and then estimates whether a gain of the surround audio channel and the central audio channel saturate or not. When the gain is saturated, the procedure enters a step for adjusting a gain of dynamic range control; while the gain is not saturated, the procedure enters a saturation process. After the step of adjusting a gain of dynamic range control, then processes a gain calculation; and then changes the gain and estimates whether the gain reaches a default value or not. When the gain reaches the default value, then the gain will be outputted. When the gain does not reach the default value, continues to process the gain calculation and change the gain.