HK1199978B - Bass enhancement system - Google Patents

Description

Bass enhancement system

RELATED APPLICATIONS

This application is a non-provisional application of U.S. provisional application No.61/580,448, filed 12/27/2011, the entire disclosure of which is incorporated herein by reference.

Background

The audio and multimedia industry is constantly striving to overcome the disadvantages of reproduced sound. For example, it is often difficult to sufficiently reproduce low-frequency sounds such as bass sounds. Various conventional approaches to improving the output of low frequency sound include using higher quality speakers with larger cone regions, larger magnets, larger housings, or larger cone excursion capabilities. Furthermore, conventional systems attempt to reproduce low frequency sound using a resonating chamber and horn that match the acoustic impedance of the microphone to the acoustic impedance of the free space surrounding the microphone.

However, not all audio systems may simply use more expensive or powerful speakers to reproduce low frequency sound. For example, some sound systems (e.g., cell phone speakers and other consumer electronic devices) rely on small loudspeakers. Furthermore, many audio systems use less accurate speakers in order to save costs. Such speakers typically do not have the ability to properly reproduce low frequency sounds, and therefore, typically the sound is not as robust or pleasing as a system that more accurately reproduces low frequency sounds.

Disclosure of Invention

For purposes of summarizing the disclosure, certain aspects, advantages, and novel features of the invention have been described herein. It is to be understood that not necessarily all such advantages may be achieved in accordance with any particular embodiment of the invention disclosed herein. Thus, the invention disclosed herein may be implemented or performed in the following manner: one advantage or group of advantages as taught herein is achieved or optimized without necessarily achieving other advantages as may be taught or suggested herein.

In a particular embodiment, a system for enhancing bass audio includes a bass enhancer having one or more processors. The bass enhancer may generate harmonics of one or more bass frequencies of the input audio signal based at least in part on available headroom in the input audio signal. In addition, the system may include an equalizer that may emphasize frequencies in the input audio signal including the lowest reproducible frequencies of the speakers. Further, the system may include a level adjuster (leveladjust) that may adaptively apply a gain to at least a lower frequency band of frequencies in the input audio signal. Such a gain may depend on the available headroom in the input audio signal.

The system of the preceding paragraph can also include any combination of the following features described in this paragraph among the features described herein. In one embodiment, the bass enhancer may generate harmonics by at least: determining an available headroom in the input audio signal; a second gain is applied to approximately half of the input audio signal. The second gain may be greater than the available headroom in the input audio signal to produce harmonics of one or more fundamental bass frequencies in the input audio signal. The bass enhancer may also include a loudness filter that may emphasize one or more bass frequencies relative to other frequencies in the input audio signal. The loudness filter may apply an anti-equal loudness curve to the input audio signal. The bass enhancer may also include an early reflection module that may filter the input audio signal with a tapped delay line. The tapped delay line may simulate reverberation reproduced by bass frequencies. The early reflection module may randomize one or both of the tapped delay and the tapped delay line coefficients in time. The system may further include a combiner that may combine the output of the bass enhancer with the input audio signal to produce a combiner output. The combiner may provide the combiner output to the equalizer. In addition, the level adjuster may further include a high-pass ramp filter, which may restore the balance of the high frequency band in the input audio signal.

In various embodiments, a method for enhancing audio may comprise: generating, with one or more processors, a harmonic of a first frequency of an input audio signal to produce an enhanced audio signal; emphasizing a second frequency in the enhanced audio signal with an equalization filter to produce an output audio signal; and providing the output audio signal to a speaker. For example, the second frequency may correspond to a speaker size setting for a speaker, as described below.

The method of the preceding paragraph can also include any combination of the following features described in this paragraph among the features described herein. For example, the second frequency may include at least some of the first frequency. The second frequency may also include a frequency band of frequencies around the speaker size setting. The first frequency may comprise bass frequencies. Generating the harmonics may include: the available headroom in the input audio signal is determined and a gain is applied to approximately half of the input audio signal. The gain may be greater than the available headroom of the input audio signal to produce harmonics of one or more fundamental bass frequencies in the input audio signal. The method may further include filtering the input audio signal with a tapped delay line that may simulate reverberation reproduced by bass frequencies. The method may further include randomizing one or both of the tapped delay and the tapped delay line coefficients in time. The input audio signal may comprise a downmix of two or more input signals. The two or more input signals may include two or more of the following signals: left front signal, right front signal, center signal, left surround signal, and right surround signal.

In some embodiments, the non-transitory physical electronic storage may include instructions stored thereon, which when executed by the one or more processors, cause the one or more processors to implement operations for enhancing bass audio. These operations may include: receiving an input audio signal; determining an available headroom in the input audio signal; and applying a gain to approximately half of the input audio signal to produce a partially clipped audio signal. The gain may be greater than the available headroom in the input audio signal, thereby generating harmonics of one or more bass frequencies in the input audio signal.

The operations of the preceding paragraphs may also include any combination of the following features described in this paragraph among the features described herein. For example, the operations may further include: the input audio signal is summed with the partially clipped audio signal to produce a combined audio signal, emphasizing frequency bands of frequencies in the combined audio signal. The frequency band of frequencies may be related to speaker size settings. The operations may also include filtering the input audio signal with a tapped delay line, which may simulate reverberation reproduced by bass frequencies. The operations may also include randomizing one or both of the tapped delay and the tapped delay line coefficients in time. Further, the non-transitory physical electronic storage device may be combined with one or more processors, memories, and/or other computing hardware components.

Various embodiments of a system for enhancing bass audio may include one or more processors that may: accessing speaker size settings input by a user; configuring a plurality of bass enhancement parameters for bass enhancement based at least in part on the speaker size settings; and applying bass enhancement to the audio input signal using the bass enhancement parameters to enhance bass frequencies of the audio input signal.

The system of the preceding paragraph can also include any combination of the following features described in this paragraph among the features described herein. For example, the bass enhancement parameters may include one or more of the following parameters: cutoff frequency, gain and bandwidth. The bass enhancement parameters may also include a cut-off frequency of a low-pass filter that may attenuate frequencies above the speaker size setting. The bass enhancement parameters may also include a bandwidth of an equalization filter that may emphasize a frequency band of frequencies in the audio input signal.

Drawings

Throughout the drawings, reference numerals may be used again to indicate correspondence between referenced elements. The drawings are provided to illustrate embodiments of the invention described herein and not to limit the scope of the invention.

Fig. 1 shows an embodiment of a bass enhancement system.

Fig. 2 shows an embodiment of a bass enhancer that may be implemented by the bass enhancement system of fig. 1.

Fig. 3 shows an embodiment of an equalizer that may be implemented by the bass enhancement system of fig. 1.

Fig. 4 shows an embodiment of a level adjuster that may be implemented by the bass enhancement system of fig. 1.

Fig. 5 shows an example frequency response curve for a loudness filter.

Fig. 6 shows another embodiment of a bass enhancement system.

Fig. 7 shows an embodiment of a downmix bass enhancement system.

Fig. 8-19 depict example output curves associated with any of the embodiments of the bass enhancement system described herein.

Fig. 20 depicts an example gain curve that may be implemented by any of the bass enhancement systems described herein.

Fig. 21A and 21B depict example time domain plots associated with an early reflection filter.

Fig. 22 depicts an example plot comparing the parallel application of two equalization filters to the serial application of the filters.

Fig. 23 and 24 depict example user interfaces for adjusting the settings of any of the bass enhancement systems described herein.

Detailed Description

Some audio systems attempt to compensate for the poor reproduction performance of low frequency sounds by amplifying the low frequency signal before inputting the signal into a loudspeaker. Amplifying the low frequency signal sends more energy to the microphone, which in turn drives the speaker with more force. However, such attempts to amplify low frequency signals may result in overdriving the loudspeaker. Unfortunately, overdriving the microphone increases background noise, causes disturbing distortions, and damages the microphone.

This disclosure describes embodiments of bass enhancement systems capable of providing an enhanced bass effect to speakers, including relatively small speakers. The bass enhancement system may apply one or more bass enhancements to the input audio signal. For example, in certain embodiments, the bass enhancement system may exploit how human ears process overtones and harmonics of low-frequency sounds to produce a perception that non-existent (or attenuated) low-frequency sounds are being emitted from a loudspeaker. In one embodiment, the bass enhancement system may generate at least a portion of harmonics of the low-frequency fundamental frequency. Playback of at least a portion of the harmonics of the low frequency fundamental frequency may result in the listener perceiving the playback of the low frequency fundamental frequency. Advantageously, in particular embodiments, the bass enhancement system may generate these harmonics without performing processing-intensive pitch detection techniques or the like to identify the fundamental frequencies.

The bass enhancement system described herein may be implemented in any computing device or device with one or more processors, for example, some examples include cell phones, smart phones, Personal Digital Assistants (PDAs), tablets, mini-tablets, laptops, desktop computers, televisions, Digital Video Recorders (DVRs), set-top boxes, media servers, audio-visual (a/V) receivers, video gamesGame system, high definition disc player (e.g., Blu-ray)Player), soundbar, and vehicle audio system.

I. Overview of Bass enhancement System

Fig. 1 shows an example embodiment of a bass enhancement system 100. The bass enhancement system 100 may be used to enhance bass sounds in devices having small loudspeakers that are unable to reproduce low frequencies or reproduce such frequencies very poorly. In some embodiments, the bass enhancement system 100 may also be used to enhance bass responses reproduced by any speaker, including speakers capable of reproducing bass frequencies.

The bass enhancement system 100 may be implemented as electronic or computing hardware, e.g., one or more processors. Examples of such hardware are described below. Further, the bass enhancement system 100 may be implemented as software or firmware, a combination of hardware and software/firmware. For example, the blocks shown in fig. 1 and subsequent figures may represent software, firmware, digital or analog hardware, combinations thereof, and so forth.

In the illustrated embodiment, the bass enhancement system 100 includes a bass enhancer 110, an equalizer 120, and a level adjuster 130. An input audio signal is received by the bass enhancement system 100 and provided to the bass enhancer 110 and the equalizer 120. Such an input audio signal may have one, two or more audio channels. For example, the input audio signals may include a pair of stereo signals, surround sound signals (e.g., 5.1, 6.1, 7.1, etc.), circular surround encoded audio signals or other matrix encoded audio, and so forth. The input audio signal may be streaming audio received over a network or audio stored on a non-transitory computer readable storage medium (e.g., CD, DVD, blu-ray disc, hard disk, etc.). For ease of illustration, however, the bass enhancement system 100 is described primarily in the context of a single audio signal (e.g., mono). Unless explicitly stated otherwise, it is understood that the features described herein may be similarly implemented in multiple channels.

With continued reference to fig. 1, the input audio signal received by the bass enhancer 110 is processed to produce a bass enhanced audio signal. Bass enhancer 1110 may use any of a variety of bass enhancement techniques. For example, in certain embodiments, the bass enhancer 110 may exploit how human ears process overtones and harmonics of low-frequency sounds to produce a perception that non-existent (or attenuated) low-frequency sounds are being emitted from loudspeakers. In one embodiment, the bass enhancer 110 may generate at least a portion of the harmonics of the low frequency fundamental frequency. Playback of at least a portion of the harmonics of the low frequency fundamental frequency may result in the listener perceiving the playback of the low frequency fundamental frequency. Advantageously, as described in more detail below (with reference to FIG. 2), the bass enhancer 110 may generate these harmonics without performing processing-intensive pitch detection techniques or the like to identify the fundamental frequencies.

In addition to or instead of performing harmonic generation, in some embodiments, bass enhancer 110 generates or simulates early reflections or reverberation of bass frequencies. This early reflection can simulate a real reflection from a wall by bass sounds. Playback of early reflections may produce a perception of deeper or richer bass content. The early reflections are described in more detail below with reference to fig. 2. In addition, in addition to or in lieu of these techniques, the bass enhancer 110 may also increase the loudness of bass frequencies by applying an inverse loudness curve filter to at least the bass frequencies.

Equalizer 120 also receives an input signal. In some embodiments, equalizer 120 emphasizes frequencies in a low frequency region around the lower reproducible limits of the speaker. Typical speakers (or speakers with their enclosures) have a low cut-off frequency that is related to the size of the speaker below which the speaker does not produce audible sound (or produces attenuated sound). The equalizer 120 may emphasize frequencies near the lower cut-off frequency, thereby enhancing the perception of bass enhancement. In some embodiments, the cutoff frequency is not a-3 dB cutoff frequency, which may be heuristically detected as detailed below. The signal path or any portion thereof from the equalizer 120 input to the equalizer 120 output is sometimes referred to herein as a time gain path.

In some embodiments, the equalizer 120 also receives input from the bass enhancer 110. The equalizer 120 may provide an output that is summed with the output of the bass enhancer 110 at summing block 112 to produce a combined bass enhanced signal. In the illustrated embodiment, the output of the summing block 112 is provided to a level adjuster 130. The level adjuster 130 may adaptively dynamically adjust one or more gains applied to the combined bass enhancement signal to vary headroom (headroom) in the audio signal over time. For example, the level adjuster 130 may dynamically compensate, at least in part, for headroom-related gain manipulation performed by the bass enhancer 110 and/or the equalizer 120 by increasing and/or decreasing the gain of the audio signal. The level adjuster is described in more detail below with reference to fig. 4. It should be noted that in some embodiments, the bass enhancer 110, equalizer 120, or level adjuster 130 may be omitted from the bass enhancement system 100 while still providing at least some of the advantages of the bass enhancement system. Further example modifications to the illustrated bass enhancement system 100 are described below with reference to fig. 6 and 7.

Fig. 2 shows a more specific embodiment of the bass enhancer 110 described above, namely a bass enhancer 210. Bass enhancer 210 may have some or all of the functionality of bass enhancer 210 described above, as well as the additional functionality shown. Embodiments of bass enhancer 210 may include any subset of the features shown in fig. 2. Some embodiments of bass enhancer 210 may also include additional features.

Bass enhancer 210 receives the input audio signal described above. In the depicted embodiment, the input audio signal is provided to a Low Pass Filter (LPF) 212. LPF212 may pass low frequencies and may attenuate frequencies greater than a cutoff frequency (Fc). The cutoff frequency may depend on a speaker size setting, which may represent the cutoff frequency of the speaker. However, in other embodiments, the cut-off frequency is user adjustable and does not necessarily depend on the speaker size. Applying a low pass filter to the input frequency may cause harmonics of the low frequency signal (rather than the intermediate or high frequency signal) to be generated (see block 214, discussed below). Harmonics in the mid-or high-frequency range may be considered as unwanted noise.

In particular embodiments, the speaker size setting may be related to the true cut-off frequency of the speaker (or the frequency response of the speaker), or may actually be the true cut-off frequency of the speaker. For example, the speaker size setting may be a frequency at which the speaker has a-3 dB or-6 dB response, or a half power response, etc. However, since it is possible to measure the speaker size setting with the speaker mounted in its housing (e.g., a television), the speaker size setting may also be a frequency different from the cutoff frequency of the speaker itself. The frequency response of the speaker may be affected by any cabinet or enclosure in which the speaker is located, and thus, in some embodiments, the speaker sizing may take into account the effect of the speaker enclosure.

The speaker size setting may be measured automatically by a processor or manually by a field engineer or other audio professional. The field engineer may heuristically establish speaker size settings for a given speaker or set of speakers (e.g., in a television), using, for example, the following process. First, the field engineer may optionally adjust the bass enhancement system 100 to provide maximum or high bass enhancement to promote easier listening discrimination of bass enhancement differences between different speaker size settings. Thereafter, the field engineer may adjust the speaker sizing settings until a balance is achieved between the quality of the bass boost and the amount of bass boost. In some embodiments, the higher the speaker size setting, the more the bass boost volume may be increased. However, the trade-off is that the quality of bass-enhanced sound may be attenuated as the size of the speaker increases. Thus, the field engineer can set the speaker size until a good balance between quality and quantity is found. One option to do this is to start with a lower speaker size setting, increase the speaker size until the field engineer hears more bass, optionally using several different test tracks to assess the bass effect. The field engineer may use a user interface (e.g., the user interface described below with reference to fig. 23 and 24) to tune the bass enhancement system in the television or other device implementing the bass enhancement system. In one embodiment, a field engineer may communicate bass boost setting changes to a device implementing a bass boost system using techniques such as described in U.S. patent application No.13/592,182 entitled "Audio adaptation system," filed 4/22/2012, the entire disclosure of which is incorporated herein by reference.

With continued reference to fig. 2, in the depicted embodiment, LPF212 provides an output signal to harmonic generator 214 that is also a separate output signal that may optionally be used by equalizer 120 (see fig. 3). In a particular embodiment, the harmonic generator 214 generates harmonics of at least a portion of the frequencies in the low-pass filtered signal. Advantageously, the harmonic generator 214 can generate these harmonics without using complex algorithms for detecting pitch or fundamental frequencies. In one embodiment, harmonics generator 214 generates harmonics by clipping (clip) at least a portion of the audio signal. Unlike some algorithms that clip both the positive and negative tracks of an audio signal and then rectify the signal to generate harmonics, in one embodiment, the harmonic generator 214 clips half of the signal. For example, the harmonic generator 214 may clip positive peaks of the audio signal, leaving negative peaks intact (or conversely, only clip negative peaks). Clipping only positive peaks (or clipping only negative peaks) may induce both odd and even harmonics, while clipping both positive and negative peaks may only cause odd harmonics (rectification is typically subsequently used to generate even harmonics from a fully clipped signal). In some instances, less processing is performed by clipping half of the signal and avoiding having to subsequently rectify the signal. In a particular embodiment, the fundamental frequency is retained in the signal.

One advantageous method for clipping a signal that may be used by the harmonic generator 214 embodiment is to clip the signal based on available headroom in the signal. For example, in one embodiment, the harmonic generator 214 calculates how much headroom the signal has, and applies a corresponding gain to the positive samples that is greater than the available headroom to cause clipping, which produces harmonics of at least a portion of the input signal frequency. Harmonic generator 214 may then apply the inverse of the gain to the positive samples to bring the samples back to their previous levels (except now that they are clipped). Using available headroom to determine clipping may be advantageous because the gain used to generate clipping may be dynamically adjusted based on the available headroom. Thus, due to this dynamic analysis based on available headroom, the harmonic generator 214 can also induce clipping regardless of the level of the input signal.

The amount of gain applied to generate clipping may be a preset amount and/or may be user defined (e.g., by a field application engineer, manufacturer, end user, etc.). In one embodiment, the gain value may be selected to attempt to amplify the signal to about 30% above full scale (e.g., about 30% above 0 dB). For example, the harmonic generator 214 may calculate that the available headroom in the signal is 10dB (e.g., by determining that the signal peaks at-10 dB in a given sample block). The harmonic generator 214 may then apply a gain of about 30% over the available headroom, or about 13dB, to the signal to induce clipping. Other gain values may be selected, such as 10% above full scale, 20% above full scale, or some other value. User adjustable control ("harmonic clipping constant") for adjusting such gain is described below with reference to fig. 24.

One useful byproduct of this half-wave clipping method for generating harmonics is that the harmonic generator 214 can also generate sub-harmonics as a side effect of the nonlinear distortion imposed by clipping half of the signal. These sub-harmonics are generated due to the inter-modulation of harmonics generated from multiple tones (tones), resulting in a richer bass sound.

In the depicted embodiment, the output of the harmonic generator 214 is provided to a loudness filter 216. The loudness filter 216 may apply an anti-equal loudness contour filter to the output of the harmonic generator 214 to increase the loudness of low frequencies in the audio signal. Fig. 5 shows an example frequency response of such a loudness filter 216. As shown, the magnitude response of the filter emphasizes lower frequencies (shown on a normalized frequency scale) relative to higher frequencies. Indeed, in some embodiments, loudness filter 216 emphasizes both frequencies below the speaker size setting and frequencies above the speaker size setting (including harmonics and sub-harmonics generated by harmonic generator 214), and optionally the fundamental bass frequencies. As described above, frequencies below the speaker size setting may be non-reproducible by the speaker or reproducible at attenuation levels. Therefore, it may be counterintuitive to emphasize these frequencies. However, in some embodiments, at least a portion of the additional bass effect may be achieved by doing so. The extended bass control shown may be exposed to control the gain of the loudness filter 216 by a user (e.g., a field application engineer, manufacturer, or end user). In some embodiments, the extended bass control may be tuned in one embodiment to obtain a maximum or maximum possible bass gain without significant distortion (or very significant distortion). Extended bass control may also be applied to equalizer 120 to further enhance bass, as described in detail below with reference to fig. 3. In alternative embodiments, loudness filter 216 may be biased so as to emphasize frequencies at, around, or above (e.g., just above) the speaker size setting.

Such an anti-equal-loudness filter may implement a weighting based on an equal loudness curve or an approximation thereof, such as an a-weighting curve, a C-weighting curve, or other equal loudness weighting curve. In one embodiment, the equal loudness filter 216 is a reverse version of one or more of the filters disclosed in Recommendation ITU-R BS.1770-2 entitled "Algorithms to measure audio program and true-peak audio level," which is published 3.2011, the disclosure of which is incorporated herein by reference in its entirety. In another embodiment, the LOUDNESS filter 216 implements an equal LOUDNESS filter (or an inverse version thereof, or an equal LOUDNESS filter-based weighting curve) according to any OF the example curves or filters described in U.S. patent application No.8,315,398 entitled "SYSTEM FOR adapting LOUDNESS SIGNALS, or LOUDNESS filters, filed on 19.12.2008, the disclosure OF which is hereby incorporated by reference in its entirety. In an embodiment, the bass enhancer 210 sets the amplitude of the loudness filter 216 based on available headroom and/or based on other characteristics of the device implementing the bass enhancement system 100 to avoid additional clipping.

In the separate processing chain of the embodiment, the input audio signal is also provided to an early reflection low pass filter 222 and an early reflection module 224. The early reflection Low Pass Filter (LPF)222 may have the same cutoff frequency as the LPF 212. Thus, in some embodiments, the LPF222 may be eliminated and the output of the LPF212 may be provided directly to the early reflection module 224. However, alternatively, in some embodiments, LPF222 may have a different cutoff frequency, which may or may not depend on the speaker size setting. Having a separate LPF222 may provide flexibility in adjusting bass boost performance.

The output of the LPF222 (or LPF212) is provided to an early reflection module 224. In certain embodiments, the early reflection module 224 may process the signal to make the low frequency signal louder by emulating the effect of placing a speaker near a wall relative to the center of the room. This spaciousness effect may enhance the volume or perception of volume in a bass response. The early reflection module 224 may achieve this effect by employing at least a tapped delay line that produces one or more delays in the signal. For example, the tapped delay line may have one, two, three, four, or more delays. In one embodiment, four delay taps may have a particularly advantageous effect. The tap coefficients may have a unity gain, or some gain other than unity gain (e.g., less than unity gain). The time domain plots of fig. 21A and 21B show time domain plots 2100, 2110 representing the pulse function and the corresponding early emissions, respectively. In the example curve 2110 of fig. 21B, it is noted that the pulse from the curve 2100 of fig. 21A is temporally reproduced along with four reflections 2112, which are attenuated and approximately negative versions of the pulse 2112.

Advantageously, in one embodiment, early reflection module 224 also enhances the perception of reflection by at least partially randomizing tap points and/or tap coefficients (e.g., tap gains). In one embodiment, the tap delays range from approximately 2ms to 48ms, and the early reflection module 224 randomly adjusts these delays in time. For example, the early reflection module 224 may randomize tap points in time (e.g., slowly) to simulate reflections of audio signals on different objects in a room. The early reflection module 224 may also randomize taps and/or coefficients differently with respect to the left and right channels (and/or left and right surround channels) to simulate different arrival times to the listener's ears. Thus, the early reflection module 224 may simulate the occurrence of actively interfering bass sound waves, while other sound waves that typically occur in a listening environment are destructive interference. Early reflections may also be used with a single speaker with the same or similar advantages.

For example, where the early reflection module 224 implements an early reflection filter with four taps, the initial tap delays may be as follows (e.g., for each channel): 2ms (tap 1), 8.33ms (tap 2), 25ms (tap 3) and 48ms (tap 4). The taps may be randomly varied in different ranges. For example, tap1 may be varied in a range of about 1ms to about 3.125ms, tap2 may be varied in a range of about 6.25ms to about 10.4ms, tap3 may be varied in a range of about 20.8ms to about 29.1ms, and tap 4 may be varied in a range of about 45ms to about 50 ms. The direction of randomization may also be random, such that some tap delays are increased while others are decreased.

While the taps may be randomly varied (or remain constant) in either the positive or negative direction, each time the taps are changed, in some embodiments a portion of the taps are increased by a random amount each time and a portion of the taps are decreased by a random amount each time until a limit (maximum or minimum) is reached. When the taps reach the limits of their range (e.g., the range described above), the next random increase or decrease may be folded in the opposite direction. For example, consider tap 2: it may be set to 8.33ms first. It may then start incrementing by a random amount (or in one embodiment, a linear amount if randomization need not be employed) until it reaches its maximum value (10.4 ms, the example range described above). Once it has reached its maximum value, the value of the tap can be folded. The folding point may be set back to the center of the tap range (about 8.33ms) and then may start to travel in the opposite direction until its minimum value is reached. Thus, after reaching the maximum of 10.4ms, tap2 can be reset to about 8.33ms, and continue to decrease to about 6.25 ms. Alternatively, the fold point may be set to a minimum (or maximum) and then begin increasing (or decreasing) again. Furthermore, the four taps may be initialized to move in opposite directions, so some will move to their high ends and some will move to their low ends.

In one embodiment, randomization occurs on a block-by-block basis such that the early reflection module 224 changes the tap delay values and/or tap coefficient values for each block of audio signal samples. For example, if the block size is 256 samples at 48kHz, the early reflection module 224 can randomize the tap delay approximately every 5.33 ms. However, the randomization frequency may be lower (e.g., every second block, every third block, etc.) or higher (e.g., randomizing multiple times per sample block). The randomization does not have to be in terms of block size. Furthermore, the randomization frequency itself may vary.

In the depicted embodiment, the output of the early reflection module 224 is provided to a multiplier block that multiplies the output by the gain setting "ER Mix". The ER Mix may be a user-set or system-defined Mix of early reflections that may control the amount of early reflections combined with the output of the loudness filter 216. In the depicted embodiment, the corresponding 1-ER Mix gain value is applied to the output of the ringing filter 216, which is combined with the early reflection output multiplied by the gain at the addition block 232. The ER Mix and 1-ER Mix gain values may be used to control the dry/wet mixing in the output audio. More reverberation from early reflections (e.g., wet sounds) can be selected by higher ER Mix gains, while less reverberant signals (e.g., dry sounds) can be selected more by lower ER Mix gains.

Thus, the multiplier block and addition block 232 may implement a convex combination of the outputs of the loudness filter 216 and the early reflection module 224, such that applying more gain to the output of the early reflection module 224 results in less gain being applied to the output of the loudness filter 216, and vice versa. These gains described herein may be adjusted by a user, which may be the manufacturer or supplier of the devices that comprise the bass enhancement system 110, a field engineer, or an end user of such devices or software. For example, another gain "C" is applied to the output of the summing block 232. As described below, this gain forms a convex combination with the output of equalizer 120 (see fig. 3).

In the depicted embodiment, an optional harmonic tail Low Pass Filter (LPF)242 is also provided. Harmonic tail LPF242 may control the amount of harmonic output of bass enhancer 210. Harmonic tail LPF242 may filter out higher order harmonics and may have a cutoff frequency that depends on the size of the speaker. In one embodiment, harmonic tail LPF242 has a cutoff frequency that is the same as or higher than the speaker cutoff frequency to which bass enhancement system 110 is applied. In one embodiment, the default value for the cutoff frequency may be about 3 times the speaker size setting, or about 2-4 times the speaker size setting. Similar to various other parameters of the bass enhancement system 100, such cutoff frequencies may be user controllable or adjustable. Higher values of the cut-off frequency may add more harmonics, resulting in richer, but possibly more distorted sound. Similarly, a lower value of the cut-off frequency may result in a cleaner but less rich sound. The output of harmonic tail LPF242 is a bass output.

Although bass enhancer 210 is described as performing a particular function, it should be understood that aspects of bass enhancer 210 may be omitted in some embodiments. For example, early reflection module 224 and associated low pass filter 222 may be omitted, or loudness filter 216 may be omitted, or harmonic tail LPF242 may be omitted, etc. Although a partial loss of bass enhancement may result, bass enhancement from other components is still advantageous. Further, it should be noted that the early reflection module 224 and/or other components of the bass enhancer 210 may be implemented independently of the algorithm used to generate the harmonics. For example, the harmonic generator 214 may use algorithms other than the described content to generate harmonics, such as by clipping the entire signal and performing rectification, using single sideband modulation, generating harmonics in the frequency domain, other techniques, or a combination of these or other techniques. Early reflection or other aspects of bass enhancer 210 may be combined with such harmonic generation techniques to produce bass enhancement.

Fig. 3 shows an embodiment of an equalizer 320. Equalizer 320 represents a more detailed embodiment of equalizer 120. As such, the equalizer 320 may have some or all of the functionality of the equalizer 120 described above. The equalizer 320 may be implemented as hardware and/or software.

The equalizer 320 receives the input audio signal described above with reference to fig. 1. The input audio signal is received by a multiplication block that multiplies the input signal by a value (1/time gain) in an attempt to determine that there is sufficient headroom to subsequently apply the time gain to the signal. The output of the multiplication block is provided to an equalization filter block 312, which may implement one or more equalization filters. These equalization filters 312 may be parametric equalization filters, semi-parametric equalization filters, or other types of equalization filters, or simply one or more bandpass filters. The equalization filter may typically be a band pass filter, enabling control of one or more of its parameters, e.g., control of center frequency, gain, bandwidth, and roll-off (or slope). The equalization filter 312 may emphasize frequency regions near, around, larger than, and/or smaller than the speaker size setting, enhancing the lowest (or approximately lowest) reproducible frequency of the speaker. In addition to having a common meaning, the term "lowest reproducible frequency" as used herein may also refer to frequencies around the speaker size setting of the speaker, frequencies within a frequency band above the speaker size setting, such settings being discussed elsewhere in the text. For example, the lowest reproducible frequency associated with the speaker may include a frequency reproducible by the speaker with half or more power, or a frequency reproducible by the speaker with-3 dB (or-6 dB) or more power from the peak reproducible frequency, and so forth. Thus, the equalization filter 312 may be a band pass filter with a center frequency set for the speaker size or based on the speaker size (e.g., offset from the speaker size by a predetermined amount).

In addition to having a center frequency based on the speaker size setting, the equalization filter 312 may also have a bandwidth that is dependent on the speaker size setting. In general, as the speaker size increases (and may reach frequencies in the voice range), the bandwidth of the filter may be smaller, thus minimizing or reducing interference with voice (vocal) or content in the mid-range of frequencies. Thus, a larger speaker with a lower speaker size setting may have a higher relative bandwidth equalization filter 312 and a smaller speaker with a higher speaker size setting may have a smaller relative bandwidth equalization filter 312. Further, since the bandwidth of the equalization filter may depend on the speaker size setting, the Q factor of the equalization filter 312 may depend on the speaker size setting. An equalization filter 312 having a center frequency set based on a higher speaker size may have a higher Q factor than a filter 312 having a lower center frequency. The higher Q factor of the higher center filter 312 may achieve the following goals: the relatively lower center filter 312 reduces the large impact of the higher center filter 312 on the speech range. For example, when the speaker size is set to 80Hz, the corresponding bandwidths may be 94Hz and 114Hz, respectively. When the speaker size is set to 250Hz, the corresponding bandwidths may be 249Hz and 383Hz, respectively.

Furthermore, the bandwidth and/or gain of the first of the two filters (or two or more or all of equalization filters 312) applied to an embodiment may also be controlled by the extended bass control described above with respect to loudness filter 216. In addition to increasing the gain of loudness filter 216, the increased size of the extended bass control described above may also increase the bandwidth of one or more equalization filters 312 to emphasize more bass and surround frequencies (including harmonics and/or sub-harmonics). Conversely, a lower extended bass setting may reduce the bandwidth and/or gain of the one or more equalization filters 312.

The plurality of equalization filters 312 may be applied in series or in parallel. However, applying the filter serially may result in a higher Q filter response, with a highly localized frequency response around the speaker sizing in certain embodiments. Fig. 22 illustrates this concept. In fig. 22, curve 2200 shows the application of two equalization filters 312 in parallel in comparison to the application of the same filters 312 in series. In curve 2200, the input signal is a logarithmic scan represented by area 2210. If two band pass filters are applied in parallel and then summed together, the resulting signal is represented by area 2220. The gain around the loudspeaker size setting is shown in the region around the normalized frequency 1. Such a signal in region 2220 has gain not only around the speaker size setting, but also over the entire frequency region. To avoid such artifacts, the summing bandpass filters may be scaled on the premise that the bandpass filters are applied in parallel. This scaling may result in a signal as shown in region 2230, where the gain around the speaker size setting is significantly reduced compared to region 2220. Although the gain in region 2230 is advantageous, a higher localized gain can be achieved by applying two equalization filters in series, resulting in a gain as shown in region 2240. In region 2240, the gain of the filter is still centered around the speaker size setting, resulting in almost the same gain as in the case of the unscaled summation of the parallel filters (region 2220).

Turning again to fig. 3, the output of the equalization filter 312 is provided to a subtraction block. Such a subtraction block may optionally subtract the low pass filtered output from the output of the equalization filter 312 (see fig. 2) based on user input (LP subtraction enabled) in one embodiment. Subtracting the low pass filtered signal may reduce the original low frequency content within the equalized filtered signal, thereby enabling the bass enhanced low frequency content to replace the original low frequency content. In some cases (e.g., where speech dominates the audio signal), it is desirable to subtract out the low frequency content. Presets may be exposed to the user to switch such settings (or alternatively, apply gains to such settings). The equalized filtered signal may then be multiplied by the user gain setting (1-C) and combined with the bass output of fig. 2 at block 322. User gain settings (1-C) represent an exemplary convex combination, where gain setting C is applied to the bass output signal of fig. 2, thereby adjusting the amount of bass output relative to the equalized output included in the output signal.

Fig. 4 shows an embodiment of a level adjuster 430, which is a more specific embodiment of the level adjuster 130. The level adjuster 430 may include some or all of the features of the level adjuster 130 described above, and may be implemented in hardware and/or software. The level adjuster 430 may at least partially compensate for headroom saving gain reduction performed by the bass boosters 110, 210 and/or the equalizers 120, 320. In this way, the level adjuster 430 may restore the signal level to the previous signal level, yet provide an option for fine tuning the gain of the low frequency region and the high frequency region of the audio signal.

In the illustrated embodiment, the level adjuster 430 receives the summed output of fig. 3, and provides the summed output to the low-level protection normalization block 412 ("normalization block 412"). The normalization block 412 can calculate how much headroom is present in the summed signal. Based on the available residual headroom, the normalization block 412 may emphasize the low level of the signal by applying a corresponding amount of gain to the signal. When a gain is applied to a low level of a signal, rather than a high level, such gain essentially performs dynamic range compression of the signal. The gain parameters and other dynamic range compression parameters applied may be user adjusted. For example, the level threshold of the knee of the dynamic range curve, the slope of the curve, and the amount of gain applied may be user-adjusted (e.g., by the end user, the manufacturer of the device implementing the level adjuster 430, a live audio engineer associated with the provider of the bass enhancement system, etc.). In an alternative embodiment, single-band dynamic range compression is applied such that the normalization block 412 increases the level of the entire signal by the same amount of gain based on the available headroom. More generally, the normalization block 412 may apply dynamic processing to the signal. Such dynamic processing may include any combination of compression, expansion, restriction, and the like, and may or may not involve the use of a fixed ratio compression scheme.

In the illustrated embodiment, the output of the normalization block 412 is provided to a multiplier that applies a gain value of 1/high-pass gain. As described below, this gain may be applied in an attempt to ensure sufficient headroom in the audio signal for subsequent gain processing. Assuming headroom is available for gain processing, the gain (or a portion thereof) can then be recovered. In the depicted embodiment, the output of the multiplier is provided to a high-pass ramping filter 414. A high pass shelving filter 414 may optionally be applied (with user adjustable gain) to boost the high frequencies so that at least some balance is restored for high frequencies, in the event that the low frequencies dominate too much. In one embodiment, the high-pass shelving filter 414 may add gain but not remove gain in low frequencies. The cut-off frequency of the tilted filter 414 may also be set according to the speaker size, or alternatively, may be set according to the low pass filter cut-off frequency described above (if different from the speaker size setting) or a different setting.

In the depicted embodiment, the output of the tilt filter 414 is provided to a protection boost block 416. The guard boost block 416 may recover the time gain and high-pass gain removed by the inverse multiplication in the multiplier of fig. 3 and 1/high-pass gain above. However, in one embodiment, protection boost 416 recovers the gain based on how much headroom is available. Protection boost 416 may recover most or all of the gain if there is sufficient headroom, and conversely, protection boost 416 may reduce the amount of gain applied if there is little or no headroom. In one embodiment, the protection boost block 416 implements a look-ahead delay line to determine what headroom is available in advance of a certain number of samples or blocks. Based on the available headroom, the protection boost block 416 may determine whether applying time and/or high-pass gain will cause clipping of the audio signal. If so, the guard boost block 416 may apply a reduced gain or no gain to the signal. Otherwise, the guard boost block 416 may apply the full gain to the signal. The guard boost block 416 may calculate one or more gains for each block of samples (or for individual samples) in the signal. In some embodiments, the guard boost block 416 uses a smoothing algorithm to the gains previously calculated block-by-block (or sample-by-sample) to smooth the gain transitions (e.g., between blocks or between samples) to avoid unwanted sound artifacts due to fast gain variations.

The output of the protection boost block 416 is provided to a High Pass Filter (HPF)418, which high pass filter 418 may optionally remove a portion of the low frequency gain in order to protect the life of the speaker. Applying too much gain to the low (or high) end of the speaker's reproducible frequency range may damage the speaker. Thus, to avoid or attempt to avoid this possibility, the high pass filter 418 may reduce the applied gain, which may be user selectable if the user desires or feels that such gain reduction is advantageous. For example, high pass filter 418 may remove or attenuate frequencies below the speaker size setting. Thus, the speaker sizing may be at or near the cutoff frequency of the high pass filter 418. Further, in one embodiment, high pass filter 418 has a steep roll-off characteristic as a result of being a higher order filter (e.g., fourth order filter). The order of the filter may also be an order other than fourth order (including lower or higher orders).

Fig. 6 shows another embodiment of a bass enhancement system 600. The bass enhancement system 600 includes many of the features of the bass enhancement system described above with reference to fig. 1-5. For example, the bass enhancement system includes a bass enhancer 610 and an equalizer 620. Although the level adjuster 130 is omitted from the bass enhancement system 600, the level adjuster 130 may be included in other embodiments. Moreover, some aspects of the bass enhancement system 600 are simplified for ease of description. For example, a cutoff frequency, an early reflection mixing coefficient, and the like depending on the speaker size are omitted. However, in various embodiments, these and other features of the bass enhancement system 100 may be implemented (or not implemented) in the bass enhancement system 600.

Advantageously, in certain embodiments, the bass enhancement system 600 may use less computing resources than the bass enhancement system 100. Since the bass enhancement system combines the input signal (via signal path 602) with the output of bass enhancer 610 and provides the combined output to equalizer 620, savings in computational resources may be realized in part. Furthermore, omitting one of the low pass filters (LPF212) of bass enhancer 210 in bass enhancer 610 reduces the use of computing resources. In contrast, bass enhancer 610 includes harmonic generator 614, harmonic tail LPF615, loudness filter 616, and early reflection LPF622 and early reflection module 624. Each of these components may have all of the functionality described above with reference to fig. 2.

In other embodiments, the early reflection LPF622 and the early reflection module 624 may be omitted in order to further reduce computational resource usage. In other embodiments, rather than providing the input signal to the harmonic generator 614, the output of the early reflection LPF622 may be provided to the harmonic generator 614 in addition to providing the output of the early reflection LPF622 to the early reflection module 624.

Fig. 7 shows an embodiment of a downmix bass enhancement system 701. The downmix bass enhancement system 701 may provide additional processing resource savings to a multi-channel environment by applying the above-described bass enhancement features to a downmix of two or more audio signals rather than to separate audio signals. In the depicted embodiment, system 701 is shown as implementing a two-channel configuration, however system 701 may also be used for more than two channels (as described below).

The system 701 includes a bass enhancement system 700, which may implement the bass enhancement system 100 or 600. Left and right input signals are received by the system 701 and provided to a combiner or summing block 702. The output of the summing block 702 is an L + R (left plus right) signal, which is provided to the bass enhancement system 700. The bass enhancement system 700 performs a portion or all of the bass processing described above with reference to systems 100 and/or 600 and provides outputs to two summing blocks 706. Similarly, the left and right input signals are each provided to a respective addition block 706. Further, the left and right input signals are each provided to a respective gain block 704, where each gain block 704 provides an output to a respective summing block 706. In an embodiment, the output of the summing block 706 is as follows:

l output ═ L input + (L + R)_{Treated by}- α (Linput + Rinput)

R output ═ R input + (L + R)_{Treated by}- β x (Ln + Rn),

wherein (L + R)_{Treated by}Is the output of the bass enhancement system 700 and α is the value of each gain block 704 in an embodiment, the values of α and β are 0.5 in different embodiments, the constants α and β may be the same or different.

In the case of more than two channels, each channel may be combined by the bass enhancement system 700 as a combined signal process. Alternatively, some channels may be processed collectively, while other channels are processed individually or not at all. For example, if the inputs include 5.1 surround sound inputs (e.g., left front, center, right front, left surround, right surround, subwoofer), the bass enhancement system 700 can enhance the combined right front and right front signals and enhance the combined left surround and right surround signals. Alternatively, the bass enhancement system 700 may enhance each of the left and right front signals separately, also enhancing the combined left and right surround signals. In yet another configuration, the bass enhancement system 700 may enhance the combined left front, center, right front, and left and right surround signals while individually enhancing the subwoofer signal. Many other variations are possible.

Example curve

In addition to the above curves, fig. 8 to 19 show additional example curves describing the scanning input and output of the above bass enhancement system. For example, FIG. 8 depicts an exemplary binaural input log scan curve 800 that may be supplied to a bass enhancement system in the time domain. Fig. 9 depicts an example binaural output log scan 900 in the time domain, corresponding to the input log scan curve 800 after processing by the bass enhancement system. Since it is a logarithmic sweep of the output, the frequency changes with time, representing only one frequency at any given point in time. The low frequencies (e.g., around the speaker size setting) are enhanced while leaving the remaining signals unaffected or less affected. Fig. 10 depicts another input log-scan curve 1000 in the frequency domain, and fig. 11 shows a corresponding output log-scan curve 1100 in the frequency domain. In the input log scan curve 1000 of fig. 10, the emphasized base bass frequencies 1010 are shown. In the output logarithmic scan curve 1100 of fig. 11, harmonics 1110 of the fundamental bass frequencies 1010 produced by the bass enhancement system are shown.

Fig. 12 depicts an input two-tone log scan curve 1200 in the time domain, and fig. 13 depicts a corresponding two-tone log scan output curve 1300 in the time domain after processing by the bass enhancement system. Similar to curve 900, curve 1300 shows how the low frequency is enhanced. Fig. 14 depicts a plot 1400 of an input two-tone log scan in the frequency domain, showing two fundamental frequencies 1410. The output two-tone logarithmic frequency sweep curve 1500 in FIG. 15 shows harmonics 1510 of these frequencies. FIG. 16 shows a plot 1600 of an input chord scan in the time domain, and FIG. 17 depicts a plot 1700 of the corresponding frequency domain showing a plurality of fundamental frequencies corresponding to the chord scan. FIG. 18 depicts a plot 1800 of the time domain output of a bass enhancement system in response to the chord scan of FIG. 16, wherein the output is enhanced. Fig. 19 shows a frequency domain output curve 1900 corresponding to the time domain output curve 1800 of fig. 18.

Fig. 20 shows a plot 2000 of an example gain curve that may be implemented in the bass enhancement system 100, e.g., implemented in any of the blocks 130, 412, 416 (see fig. 4). Although not necessarily so, the gain curve may be tuned differently for blocks 412 and 416. For example, different gain curves may have different inflection points, threshold levels, and gain setting amounts (as described with reference to fig. 4).

Further, it should be noted that any low-pass and/or high-pass filter (or other filters described herein) may have any filter order. For example, the order of the filter may be quadratic, cubic, quartic, or higher. In one embodiment, the filter order may be selectable to provide higher order filtering in a system with additional processing power to handle higher order filtering, and lower order filtering in a more resource constrained system.

Further, embodiments of the features described herein may be implemented with or in conjunction with the systems and features described in U.S. patent No.6,285,767, entitled "Low-Frequency Audio enhancement system," the disclosure of which is hereby incorporated by reference in its entirety.

Furthermore, for convenience, embodiments of the present disclosure describe applying various enhancements (e.g., gain and/or filtering) to an audio signal or an input audio signal. It should be understood that in some embodiments, after a first component described herein applies enhancement to an input audio signal, a second component may then apply gain or filtering to the enhanced input audio signal output by the first component. However, for ease of description, the present disclosure may sometimes interchangeably refer to the second component as applying enhancement to the input audio signal rather than the enhanced input audio signal. It should be noted that most, if not all, of the processing described herein may be implemented in a different temporal order than that shown and described, and thus the present description refers generally to components enhancing an input audio signal, even though in reality these components enhance a version of the input audio signal that has been enhanced by another component.

Example user interface

As described above, a field engineer, manufacturer, or end user (e.g., listener) may use a user interface to tune a bass enhancement system in a television or other device implementing a bass enhancement system (e.g., any of the bass enhancement systems described herein). Fig. 23 and 24 depict examples of such user interfaces 2300, 2400. The user interface 2400 of fig. 24 is a continuation of the user interface 2300 of fig. 23, which user interface 2400 may be reached by scrolling down from the user interface 2300. The user interfaces 2300, 2400 can be implemented in a browser or other application in addition to a browser. Further, the user interfaces 2300, 2400 may be accessed over a network or locally at a device tuned using the user interfaces 2300, 2400.

The user interfaces 2300, 2400 include a number of user interface controls 2310, 2410 that enable a user to adjust various settings or parameters of the bass enhancement system. The example user interface controls 2310, 2410 shown include checkboxes, sliders, and text boxes. These controls are merely exemplary, and other types of controls may be used to achieve the same or similar results. The following is an example summary of the setup of some aspects shown in the user interfaces 2300, 2400. Many of these arrangements are described in more detail above. The ranges set forth for these settings are exemplary only and may vary in other embodiments.

Enable/Disable: this control is used to enable and disable bass processing, including processing by the entire bass enhancement system.

HP Only Enable/Disable: if the control is enabled, only a high pass filter is applied to the signal. The cut-off frequency (Fc) of the high-pass filter can be calculated as: speaker size x high pass ratio (see below).

ELC Filter Enable/Disable: the equal loudness curve-based filter (e.g., loudness filter 216, etc.) applied to the harmonic path is enabled.

In minus LP Enable/Disable: when enabled, the unprocessed low-pass path is subtracted from the temporal gain path (e.g., equalizer 120 or 320 path) before the temporal gain path (e.g., equalizer 120 or 320 path) is mixed with the harmonic path (e.g., bass enhancer 110 path, mixing performed by mixer 112).

Input Gain (dB): the control may be utilized to change the gain of the signal prior to processing by the bass enhancement system. Since multiple audio sources may have varying levels, the control may allow for boosting very low signals or reducing very high gain signals. The control ranges from-60 dB to 0 dB.

Output Gain (dB): the output gain applied after the bass enhancement system has been processed is set. The output gain is specified in decibels, ranging from-60 dB to 0 dB.

HP Comp Speaker Size Ratio: the control sets the high-pass gain compensation cutoff frequency (Fc) to be a ratio of the speaker sizes. The high-pass compensation Fc was calculated as: HP compensates for ratio x speaker size. The range of the control is [1, 8 ].

HP Gain (dB): this control sets the high-pass gain of the high-pass shelving filter 414 applied to the signal (see fig. 4). The range of this control is 0, 18 dB.

Speaker Size (Hz): the control sets the speaker size setting. The range of the control is [40, 800] Hz, although other ranges may be used as described above.

Low Pass Speaker Size Ratio: this control is used to set the Fc of the applied low pass filter to the ratio of the speaker sizes: fc ═ low pass filter size ratio x speaker size. The range of the control is [0.5, 6 ].

Max Gain LP Only Enable/Disable: when enabled, the maximum normalization gain is applied only to the low-pass filtered signal. This gain may be implemented by the low-level protection normalization block described above, for example, the gain described above with reference to fig. 20 may be selected.

Max Norm Gain (dB): this control sets the maximum normalized Gain that can be applied to the signal (low pass or wideband signal depending on the setting of the Max Gain LP Only Enable/Disable control). The range of this control is [0, 30] dB. This gain may be implemented by the low-level protection normalization block 412 described above.

Max Norm Gain Thresh: a threshold is set for the low end of the maximum gain curve implemented by an embodiment of the low level protection normalization block 412. The range of the control is [10, 6.0].

Max Norm Gain Knee: the inflection point of the maximum gain curve implemented by the embodiment of the low-level guard normalization block 412 is set. The range of the control is [0.1, 0.6 ].

High Pass Ratio: the control sets the speaker size ratio of the high pass filter applied to the signal. The Fc of the high pass filter applied to the signal can be calculated as: high pass ratio x speaker size. The range of the control is [0.1, 1 ].

Harmonics Clip Const: the control sets the amount of gain applied in the harmonic generation path when generating harmonics as a percentage of the internally calculated available headroom. The range of the control is [1, 6 ].

Harmonics Gain (dB): the control sets the amount of gain applied to the harmonic generation path. The control range is [ -60, 24] dB. In an embodiment, 0dB is full scale, so any value above 0dB may cause clipping. In other embodiments, values below 0dB may cause clipping depending on the headroom in the audio signal.

Harmonics LPF Speaker Size Ratio: the control sets Fc of the harmonic-producing low-pass filter path (e.g., block 212 of fig. 2) to a ratio of speaker sizes. The Fc of the harmonic LPF may be calculated as: harmonic LPF speaker size ratio x speaker size. The range of the control is [0.1, 6.0].

Path MIX Const: the control sets a mixing ratio between the harmonic generation path and the time gain path. The range of the control is [0, 1 ]. The higher the setting, the more harmonic path signals are added to the mix.

Temporal Gain (dB): the control sets the time gain applied to the signal. The range of this control is [0, 24] dB.

Temporal Slope: the control sets the slope of the time gain filter applied to the signal. The range of the control is [0.25, 4 ].

Early Reflection Enable/Disable: the early reflection path added to the low pass path is enabled.

Tap1 Mix: the mixing coefficient of the first early reflection tap is set.

Tap2 Mix: the mixing coefficient of the second early reflection tap is set.

Tap3 Mix: the mixing coefficient of the third early reflection tap is set.

Early Reflection Mix: the mixing ratio of the early reflections is set.

Although these parameters may be set separately using the user interfaces (or user scripts, etc.) of fig. 23 and 24, advantageously in certain embodiments this may not be necessary. Instead of requiring the user to tune a large number of bass parameters (e.g., the parameters shown on several different types of devices), the bass enhancement system may advantageously enable the user to tune one or a small number of parameters, and then the bass enhancement system may tune several other parameters based on the user tuned parameters. For example, the speaker size settings described herein may be selected by a user. When a speaker size setting is specified, the bass enhancement system may automatically set a variety of other parameters, as described herein. For example, some or all of the following parameters may depend on the speaker size settings, including: the cutoff frequency (Fc) of the low-pass filter 212; early reflection mixing; the gain, center frequency, and/or bandwidth of the equalization filter 312; the gain and/or Fc of the high-pass shelving filter 414; fc of high pass filter 418; and the gain of low pass filter 242. Similarly, as described above, the extended bass control described herein may be user-adjustable, and the resulting extended bass control may affect the Fc and/or gain of loudness filter 216 and the gain, center frequency, and/or bandwidth of equalization filter 312. Similarly, the temporal gain control described herein may be user adjustable and affect the gain, center frequency, and/or bandwidth of the equalization filter 312 and the gain and Fc of the high-pass shelving filter 414.

Thus, once a user has entered a desired speaker size setting, extended bass control, and/or time gain control, the bass enhancement system can set a variety of other parameters that facilitate rapid tuning of a plurality of different devices and that enable the bass enhancement system to effectively enhance the bass of the plurality of different devices.

Additional examples

Although the bass enhancement system described herein may provide improved bass in a variety of devices, in some devices with very small speakers, different advantages may be achieved. In particular, although bass may be enhanced to some extent, one of the many advantages of a bass enhancement system is that it can generally enhance speech by making it warmer or richer. Such an advantage is not limited to an apparatus having a very small speaker but can also be present in an apparatus having a large speaker, which also exhibits a more significant bass effect due to the bass boost system. Thus, bass enhancement systems provide the advantage of speech enhancement, enabling low bandwidth speech to sound as if there were more frequencies present. This advantage results at least in part from the addition of harmonics and sub-harmonics to the speech frequency range, which can at least partially compensate for speech frequencies lost due to limited bandwidth. Thus, bass enhancement systems may be used as speech enhancement in cell phones, fixed phones, conference calling devices, answering machines, and the like.

Further, in some embodiments, a bass enhancement system may be used to enhance frequency ranges other than bass or low frequencies. For example, a bass enhancement system may be used to emphasize any subset of frequencies within the audio spectrum, including speech frequencies above typical bass frequencies, and the like. The speaker sizing described herein may also be used to perform enhancement of the high frequency range where the speaker is also cut off. Bass enhancement systems may also be used to enhance music including bass frequencies and/or higher frequencies.

Term of V

Many other variations than those described herein will be apparent in light of this disclosure. For example, depending on the embodiment, certain acts, events or functions of any algorithm described herein can be performed in a different order, may be added, merged, or left out all together (e.g., not all described acts or events are necessary for the practice of the algorithm). Further, in some embodiments, acts or events may be performed concurrently, e.g., through multi-threaded processing, interrupt processing, or multiple processors or processor cores, or on other parallel architectures, rather than sequentially. Further, different tasks or processes may be performed by different machines and/or computing systems that are capable of working together.

The various illustrative logical blocks, modules, and algorithm steps described in connection with the embodiments disclosed herein may be implemented as electronic hardware, computer software, or combinations of both. To clearly illustrate this interchangeability of hardware and software, various illustrative components, blocks, modules, and steps have been described above generally in terms of their functionality. Whether such functionality is implemented as hardware or software depends upon the particular application and design constraints imposed on the overall system. For example, the vehicle management system 110 or 210 may be implemented as one or more computer systems or as a computer system including one or more processors. The described functionality may be implemented in varying ways for each particular application, but such implementation decisions should not be interpreted as causing a departure from the scope of the present disclosure.

The various illustrative logical blocks and modules described in connection with the embodiments disclosed herein may be implemented or performed with a machine having: a general purpose processor, a Digital Signal Processor (DSP), an Application Specific Integrated Circuit (ASIC), a Field Programmable Gate Array (FPGA) or other programmable logic device, discrete gate or transistor logic, discrete hardware components, or any combination thereof designed to perform the functions described herein. A general purpose processor may be a microprocessor, but in the alternative, the processor may be a controller, microcontroller, or state machine, combinations of the above, or the like. A processor may also be implemented as a combination of computing devices (e.g., a combination of a DSP and a microprocessor), a plurality of microprocessors, one or more microprocessors in conjunction with a DSP core, or any other such configuration. The computing environment may include any type of computer system, including but not limited to microprocessor-based computer systems, mainframe computers, digital signal processors, portable computing devices, personal organizers, device controllers, and computing engines within the device, to name a few.

The steps of a method, process, or algorithm described in connection with the embodiments disclosed herein may be embodied directly in hardware, in a software module executed by a processor, or in a combination of the two. A software module may reside in RAM memory, flash memory, ROM memory, EPROM memory, EEPROM memory, registers, hard disk, a removable disk, a CD-ROM, or any other form of computer-readable storage medium known in the art. An example storage medium may be coupled to the processor such that the processor can read information from, and write information to, the storage medium. In the alternative, the storage medium may be integral to the processor. The processor and the storage medium may reside in an ASIC. The ASIC may reside in a user terminal. In the alternative, the processor and the storage medium may reside as discrete components in a user terminal.

Conditional language, as used herein, wherein e.g., "can," "might," "etc," is generally intended to mean that some features, elements, and/or states included in some embodiments but not included in other embodiments unless specifically stated or otherwise understood in context when used. Thus, such conditional language is not generally intended to imply that it is in any way required for one or more embodiments, or that one or more embodiments necessarily include the following logic, with or without author input or prompting: to determine whether such features, elements and/or states should be included or performed in any particular embodiment. The terms "comprising," "including," "having," and the like are synonymous and are used inclusively, in an open-ended fashion, and do not exclude additional elements, features, acts, operations, and the like. The synonymous term "or" is used in its inclusive sense (and not in its exclusive sense) such that when used, for example, to connect a list of elements, the term "or" means one, some or all of the elements in the list.

While the above detailed description has shown, described, and pointed out novel features as applied to various embodiments, it will be understood that various omissions, substitutions, and changes in the form and details of the device or algorithm illustrated may be made without departing from the spirit of the disclosure. It is to be understood that some embodiments of the invention described herein may be implemented in a form that does not provide all of the features and benefits described herein, as some features may be used or practiced separately from others.

Claims (5)

1. A system for enhancing bass audio, the system comprising:

a bass enhancer configured to receive an input audio signal and comprising:

a low pass filter configured to filter an input audio signal to produce a low pass filtered signal,

a harmonic generation module configured to generate harmonics of one or more bass frequencies of the low-pass filtered signal based at least in part on available headroom in the low-pass filtered signal, thereby generating a harmonic signal, an

An early reflection module configured to filter an input audio signal with a tapped delay line having one or more tapped delays to generate an early reflection signal, wherein the early reflection module is further configured to randomize in time one or both of: the one or more tapped delays of the tapped delay line, and coefficients of the tapped delay line;

a first combiner configured to combine the harmonic signal with the early reflection signal to produce a bass enhancement signal;

an equalizer configured to receive the input audio signal and emphasize frequencies of the input audio signal including a lowest reproducible frequency of a speaker to generate an equalized signal;

a second combiner configured to combine the bass enhancement signal with the equalized signal to produce a combined bass enhancement signal; and

a level adjuster configured to adaptively apply a gain to at least a lower frequency band in the combined bass enhanced signal, and to adaptively adjust the gain applied to the combined bass enhanced signal, wherein the gain is dependent on available headroom in the low-pass filtered signal.

2. The system of claim 1, wherein the bass enhancer is further configured to generate harmonics by at least:

determining an available headroom in the low-pass filtered signal; and

a second gain is applied to approximately half of the low-pass filtered signal, the second gain being greater than the available headroom in the low-pass filtered signal, thereby generating harmonics of one or more fundamental bass frequencies in the low-pass filtered signal.

3. The system of claim 1 or 2, wherein the bass enhancer further comprises: a loudness filter configured to emphasize the one or more bass frequencies relative to other frequencies in the harmonic signal.

4. The system of claim 3, wherein the loudness filter applies an anti-equal loudness curve to harmonic signals.

5. The system of claim 1, wherein the tapped-delay line is configured to simulate reverberation reproduced by low audio frequencies.