TWI491277B - Dynamic volume control and multi-spatial processing protection - Google Patents

Dynamic volume control and multi-spatial processing protection Download PDF

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TWI491277B
TWI491277B TW098138834A TW98138834A TWI491277B TW I491277 B TWI491277 B TW I491277B TW 098138834 A TW098138834 A TW 098138834A TW 98138834 A TW98138834 A TW 98138834A TW I491277 B TWI491277 B TW I491277B
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signal
level
system
left
difference
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TW098138834A
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TW201119421A (en
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Christopher M Hanna
Gregory Benulis
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That Corp
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Description

Dynamic volume control and multi-space processing protection (2)

This application is related to US Provisional Patent Application US 61/114,684 (filed on November 14, 2008 by Christopher M. Hanna, Gregory Benulis and Scott Skinner) and US 61/114,777 (on November 14, 2008). The application was filed by Christopher M. Hanna and Gregory Benulis; the two applications were referenced in this case. This application also has a US interim patent application in the joint trial.      (Attorney's Docket No. 56233-427-THAT-26), presented concurrently with Christopher M. Hanna, Gregory Benulis, and Scott Skinner; and assigned to the assignee.

This application is related to audio signal processing, especially the volume control of audio signals and multi-space processing protection.

When watching TV, the volume change can be irritating and often requires the viewer to manually adjust the volume. As an example, changes in the perceived volume tend to occur when switching TV channels. Another example is the perceived volume change that can occur between a television program and a commercial. These large relative changes are generally attributed to the lack of level control at the broadcast point or the lack of signal compression introduced at the time of production. The loudness of perception increases, and some unknown reasons are multi-space processing. The surround space effect (analog surround) of the two-channel system is introduced when the studio performs sound processing in certain program materials. If this type of broadcast sound is processed on TV to introduce a two-channel surround effect (as is currently the case with many TV models), the level of perception may change dramatically. Additional spatial processing can cause the central image (usually a dialogue) to be almost incomprehensible. In all cases, the automatic volume control technology minimizes the discomfort of the listener and maintains a consistent volume level. When concentrating on adjusting the volume level of the broadcast point, it seems that the mitigation of this problem is less helpful. In fact, with the evolution of high dynamic range digital television (DTV) broadcasting, television viewers are now aware of the widest differences in loudness.

According to one aspect of the disclosed system and method, a system provides dynamic control of a perceived volume of a stereo program including left and right channel signals, including: a dynamic volume controller configured and configured to maintain the stereo program in perception Constant volume level; and an over-space processing protection processor, the composition and configuration of which is used to control the difference signal level generated by the function of the left channel signal minus the right channel signal (LR). Compared with the sum signal level generated by the function of the right channel signal plus the left channel signal; and the excess space processing protects the processor for sound signal processing, the LR is controlled. Enhanced.

According to another aspect, a system provides dynamic control of a perceived volume of a stereo signal program including the left and right channels, including: a dynamic volume controller configured and arranged to maintain a constant volume level of the stereo program at a perceived level; The program conversion detector is configured and configured to provide a program switching signal, indicating that the left channel and the right channel signal have been reduced below a threshold level for at least one valve period, thereby contributing to the sound level of the left and right channel signals. There is a possible change; and the dynamic volume control responds to the program conversion signal.

According to still another aspect, a system provides dynamic control of a perceived volume of a stereo signal program including the left and right channels, and includes: a dynamic volume controller configured and configured to maintain a constant volume level of the stereo program at a perceptual level The dynamic volume controller includes at least a compressor that responds to the high and low rise and release thresholds, and defines the perceived soft, normal, and loud volume levels.

Still in accordance with yet another aspect, a system provides dynamic control of the perceived volume of a stereo signal program including the left and right channels, including: an oversized spatial processing protection processor, the composition and configuration of which is controlled by the left channel signal minus The difference signal level generated by the right channel signal (LR); and the contour filter for forming a difference signal.

According to still another aspect, a system provides for dynamically controlling the perceived volume of a stereo signal program including the left and right channels. The system includes: an excess spatial processing protection processor, the composition and configuration of which is used to control the difference signal level generated by the left channel signal minus the right channel signal (LR); and the difference signal can be formed. A contour filter.

Now explore the diagram of the implementation. Other embodiments are additionally used or substituted for use. Obviously known or unnecessary details may be omitted to save space or make the description more effective. Conversely, the implementation of certain embodiments may not necessarily have all the details disclosed.

Dynamic Volume Control (DVC) System

Description A DVC system is used to dynamically control the volume of an audio signal. The system, its composition and configuration mode can be dynamically manipulated and the volume is modified when an emergency occurs. Embodiments described herein are constructed and configured to maintain a sound frequency band for a constant volume level of perception. The DVC system can be fully digital and can be cost effective in software (C, combined language, etc.) or digital hardware (HDL description), although the system should be a completely analog or mixed analog digital system. Market applications include television sound, DVD player sound, set-top box (STB) sound, radio sound, and other highly fifi (hifi) and non-hifi (non-hifi) audio products. If there is no DVC system of the type described herein, the perceived volume level may change drastically when a broadcast or source of program material changes, or when the sound broadcast or source changes. These volume changes can be irritating and often require manual volume adjustment by the listener. A specific example is the volume change that occurs when a TV channel is converted. Another example is the volume change between a TV show and a TV commercial. In these two examples, the DVC system eliminates the discomfort of the listener and maintains a more consistent volume level.

FIG. 1 shows an embodiment of such a DVC system 100. The system 100 receives two input signals: a left channel signal L at input 102, and a right channel signal at input 104. The DVC system configuration in this described embodiment is based on a digital implementation of a classical compressor design (THAT Design Note 118), which is only possible with flexibility and additional modifications in digital implementations. The system 100 includes an RMS level detector 110 for providing a signal indicative of the sum of the RMS averages of the left and right channel signals L and R, a log conversion block 112, and a signal average AVG block 114. The logarithmic transform block 112 transforms the output of the RMS level detector 110 from a linear domain to a logarithmic domain. System 100 responds to a number of control signals indicating whether a condition exists and must be replied from the system. The system 100 also includes a main processor (not shown) that is comprised and configured to perform the operations of the DVC system 100. The illustrated embodiment responds to some control signals, including: a target level signal is provided by the target signal generator 116, and an up threshold signal is generated by the rising threshold signal device 118, and a release threshold is generated. (not shown), a gateway threshold signal is generated by the gate threshold signal device 120, a rising ratio threshold (not shown), a release ratio threshold (not shown), a proportional signal The proportional signal device 122 generates a mute lock signal generated by the mute lock device 124 in response to a program change detector (PCD; not shown). The devices 116, 118, 120, 122 may be only controllers that are user adjustable. Device 124 can be configured to receive a signal from a controller when the television channel is switched, or from a silence detector (not shown) when both inputs 102 and 104 have been muted. The target level signal 116 indicates its level in decibels (dB) relative to the full scale input, ie, the target volume. The rising threshold value 118 indicates that REF must exceed the dB of the AVG value until the attack time is reduced by a factor of N (N can be any number); in an embodiment, N=10. The release valve limit signal indicates that REF must be lower than the AVG value in dB before the release time is reduced by M times (M can be any number); and in an embodiment, M=10. The gateway threshold value 120 indicates that REF can be lower than the AVG value (negative dB number) before all left and right channel gain adjustments are locked. The rise ratio threshold value indicates that the REF may exceed the absolute amount (in dB) of the target level signal 116 before the volume controller begins to attenuate the input signal. The release ratio threshold value indicates that REF may be lower than the absolute amount (in dB) of the target level signal 116 before the volume controller begins to add gain to the input signal. The proportional signal 122 adjusts the AVG value according to the desired compression ratio.

The target level signal 116 is subtracted from the output of the logarithmic transform block 112 by the signal adder 126, thereby providing a REF signal to the signal average AVG block 114, a comparator 128 and a second comparator 130. The REF signal indicates the volume level of the input signal relative to the threshold value to be listened to. The AVG signal can be considered as an ideal gain recommendation for an instant (before the rise or release process). The output of the signal average block 114 is the AVG signal, which is a function of the average value of the REF signal. The AVG signal is applied to the signal adder adder 132 where it is added to the rising valve limit signal 118. In a similar manner (not shown), the AVG signal is combined with a release valve limit. The AVG signal is also applied to the signal adder adder 134 where it is added to the gateway threshold signal 120. The output of the signal adder 132 is applied to the rising threshold comparator 128 where it is compared with the REF signal; at this time, the output of the signal adder 134 is applied to the gate threshold comparator 130. And compare it with the REF signal there. The AVG signal is also multiplied by the signal multiplier 136 via the proportional signal 122. The output of the comparator 128 is applied to the rising/releasing selection block 138, so that an Att (rising) signal or a Rel (release) signal is provided to the signal averaging block 114, and the status of the mute locking signal 124 is viewed. And responded. The output of the release threshold AVG (not shown) is also compared to the REF signal and applied to the rising release selection block. The comparator 130 provides an output of the Hold output to the signal averaging block 114. Finally, the signal multiplier 136 provides an output to a log-to-linear signal converter 140 such that an output is provided there for each of the signal multipliers 142 and 144 where they are separately scaled. The left and right channel signals are provided by the inputs 102 and 104, thus providing modified outputs Lo and Ro for the left and right channel signals.

Referring to FIG. 1, the RMS level detector 110 senses the sound level of the input signal. It should be noted that any type of signal level detector can be used while displaying the RMS level detector. For example, a peak detector, an average detector, a perceptual-based level detector (such as the ITU1770 loudness detector or CBS loudness detector), or other detection that can be used to sense sound levels. Device. These level detectors typically have time constants that can be dynamically and independently adjusted. One method of adjusting these time constants is based on the packet or general shape of the input signal such that the time constant changes with the signal. In other embodiments, the time constant is fixed. To simplify data processing, the sound level can be converted to a logarithmic field using logarithmic conversion block 112, as shown. In a multi-band system, a separate RMS detector can be used for each frequency band. The signal average block 114 is structured and arranged to calculate an average of REF versus rise and release times. The output signal AVG of the signal average block 114 is adjusted by the multiplier 136 by the desired compression ratio to produce a gain value to be applied. Finally, the gain can be converted back to a linear domain by the logarithmic linear converter 140 to be applied to the left and right signals L and R to produce modified left and right signals Lo and Ro.

The target output level indicated by the target level signal 116 is subtracted from the sensed level at the output of the logarithmic transform block 112 to determine the difference between the true and desired sound levels. This difference indicates the level of the input signal relative to the target level signal 116, which is referred to as a reference (REF) signal. The target level signal can be a user input, such as a simple knob or other preset value, thereby controlling the desired sound level. The threshold value can be fixed or can vary depending on a function of the input signal level of the preferred compression position relative to the input dynamic range. Once the REF signal is obtained, the average block 114, the rising threshold comparator 128, and the gate threshold comparator 130 can be provided as an input. The output of the rising threshold comparator 128 is applied to the rise/release selection block 138 where it receives a MuteHold signal 124 from a program transition detector.

The gateway threshold signal 120, when added to the existing average AVG, indicates that the lowest REF value can be achieved before the left and right channel gain adjustments (142 and 144) are locked. The gateway threshold comparator 130 receives the instantaneous signal level (REF) signal and determines whether the sound level represented by REF is reduced below the certain threshold. If the instantaneous signal level (REF) is greater than the threshold value of the gateway, and the amount is lower than the average signal level (AVG) appearing at the output of block 114, the gain applied to the signal in the signal path Keep steady until the signal level rises above the threshold. The intention is to prevent the system 100 from applying more gain to very low level input signals, such as noise. In an infinitely locked system, the gain can be fixed forever, until the signal level rises. In a system with missing locks, the gain can be added slowly (far slower than the release time). In one embodiment, such a gate lockout threshold can be adjusted; in another embodiment, the threshold value set by the gate threshold 134 can be fixed.

The program transition detector or MuteHold senses when the input is "silent". When the user switches the TV (TV) channel, the sound level between the two channels may change, greatly increasing or decreasing. In general, TV manufacturers will silence the channel when it is converted, so that viewers will not be bored with the transient phenomenon of the sound. The program transition detector is designed to check for such silence by determining if the sound level is reduced below a predetermined threshold (MuteLev) for a predetermined amount of time (MuteTime). If the instantaneous sound level (REF) is below the threshold for a period of time or "silence time", the program transition can be detected. If a program conversion is detected, the speed of such rise and release times (more on this later) will increase. As the speed increases, if a loud channel is switched to a soft channel, the increased release time can increase the gain faster to match the target sound output level. Conversely, if you switch from a soft channel to a loud channel, the increased rise time allows the gain to increase more quickly to match the target. If the sound level rises above the threshold before the "Silent Time" deadline, the program transition cannot be detected. In alternative embodiments, the "silence time" and mute threshold values may be fixed, user adjustable, changed, or otherwise.

Figure 2 illustrates one embodiment of a state diagram of one of the silent detection algorithms on the program transition detector operation. The job 200 includes three states: a MUTE OFF state 202, a MUTE ON state 208, and a MUTE HOLD state 212. In the MUTE OFF state 202, the REF signal of the signal adder adder 126 can be periodically compared to the MuteLev threshold level at 204 to determine if REF>MuteLev or REF<MuteLev. If REF>MuteLev, the job stays in state 202 and continues in that state. In this state, MUTE ON=0, MUTE HOLD=0, and the rise and release times are at the normal set values. As REF<MuteLev, mutes are detected and the job transitions from 206 to state 208 MUTE ON. Once transitioning to state 208, MUTE ON = 1, and at state 208, the program transition detector determines if the silence condition has remained for a predetermined period of time. The detector transitions back to state 202 if the muting condition does not last long enough and REF>MuteOffLev occurs before the timer expires. This can happen when the show is silently paused. However, when the timer determines that MuteTime has expired, program conversion has occurred. In the state where REF>MuteOffLev returns, the detector will transition from 201 to MUTE HOLD state 212. In this state, the rise and release times are accelerated, so that the relatively loud signal turns to a soft sound, and the relatively soft signal turns to a loud sound at a predetermined time limit (MuteTime). In Figure 2, the set value displayed by the timer in state 208 will be the same as in state 212. Obviously, these settings can also be different. In state 212, if REF falls below the MuteLev setpoint before the MuteTime expires (ie, REF<MuteLev), the state transitions from 214 back to state 208. However, if MuteTime does expire, the detector will transition from 216 back to state 202.

In an embodiment, MuteTime and MuteLev are adjustable. In a given implementation, the mute time and the mute level may also be fixed. The mute threshold is set below the gate threshold. The mute detection algorithm can be used in either automatic or manual mode. In the automatic mode, the system 100 detects a mute condition upon a channel change. The program change detector can also operate in manual mode, during which a "mute" signal received by a television or other device indicates that the channel is being converted. Further, the program change detector can also receive a signal from the user's remote control to decode whether the user converts the channel. The system 100 can also operate using rise and fall thresholds. In a given time window, if the sound level jumps to the limit of its crossing threshold 118, the system 100 can operate in "fast rise" mode. In one embodiment, if REF exceeds AVG by the rising threshold, then this fast rising mode increases the rise time constant to quickly reduce the gain of this increased sound level. Similarly, if the traversing release threshold is exceeded, the system will operate in a fast release mode and the gain will increase rapidly. These rise and release time constants are independently adjustable from each other and between the high and low bands of the multi-channel system.

In some implementations, the maximum gain applied to the input signal can be limited. This limits the amount of gain that it applies to the soft segment. If the loud segment (the thunder in the movie) follows the soft segment, the unrestricted gain may cause a significant sound overshoot before the gain of the rise time decreases.

The average block 114 receives the REF, rise, release, and lock signals and determines the average (AVG) of the REF signal based on the functions of the rise, release, and lock signals. The AVG signal can then be adjusted for volume control by the compression ratio applied to the original signal. The AVG signal represents the REF signal processed by the rise or release time constant. When the REF chopping wave of the average block 114 changes, the AVG signal is affected, and the first need to be adjusted according to the desired compression ratio. It should be understood that system 100 does not compress indefinitely. Once the value of the AVG signal is adjusted according to the compression ratio, the AVG signal is multiplied by the multiplier 136 by the ratio setting means 122 by -(1 - ratio). Therefore, taking the 4:1 compression ratio as an example, the AVG signal can be multiplied by -(1-1/4) or -3/4. Therefore, if the sound is 20 dB (above the threshold), the AVG signal is equal to 20 dB (after the rise time constant has elapsed). Multiplying 20dB by -3/4 yields a value of -15dB. Therefore, the 20dB sound above the threshold is attenuated to 5dB after applying a -15dB gain. 20/5=4, this is the 4:1 compression ratio.

The compression ratio applied to the signal can be a single slope ratio. For example, a 4:1 ratio can be applied to the incoming signal, depending on the level threshold. If the AVG is above this threshold, the signal is reduced by a factor of four (in ascending ratio). Conversely, if the AVG is below the threshold, the signal is amplified four times (to release the ratio).

In another embodiment, the compression ratio may be different depending on whether the AVG signal is above or below the target level threshold provided by device 116. For example, if the AVG signal is above the target level threshold, the signal can be reduced by a factor of four (as shown in the previous example). In contrast, if the AVG signal is below the threshold, different ratios can be applied to amplify the input signal, referred to as a 1.5:1 ratio. This configuration allows the loud signal to be compressed above the proportional threshold and maintains the sound level for a soft conversation (eg whisper). The configuration described above can be regarded as a movie mode; the harsh edge of the loud signal is removed, but the soft signal (the rustling of the leaves, etc.) is maintained at the original level. This is a good mode for one of the loud volume settings. Therefore, it is expected to achieve a more complete dynamic range while still compressing loud and annoying signals. Another configuration involves heavy compression (eg, 10:1) of the AVG value above or below the level threshold. Severe compression here means "night mode" because you can listen to all the sounds of the program (both loud and soft) without having to turn the volume up (to soft) or turn down (to loud). Night mode is a good mode of low volume setting, which is often loved by viewers in the middle of the night.

Still further, another embodiment is to consider the use of high and low rise release ratio thresholds. In such an embodiment, the two thresholds define three regions of a loudness space: soft, normal, and loud. In each of the windows, different compression ratios can be applied. For example, a ratio of 1.5:1 can be used to amplify a soft signal, a ratio of 1:1 can be used to maintain a normal signal, and a ratio of 4:1 can be used to attenuate a loud signal. In this multi-window system, the original dynamic range can be more accurately maintained, while the loud and soft signals of the edges can be attenuated or amplified separately.

Finally, if the logarithmic domain is processed, the calculated compression ratio can be "linearized" at 140 before the gain is applied to the input signal.

3 shows a single channel system 300 in which one DVC system 302 can apply the same gain to the left channel (L) and right channel (R) signals. Specifically, the output of the DVC system 302 (provided by the logarithmic linear signal converter 140) is dynamically set to the gain of each of the amplifiers 308 and 310, respectively, as shown in FIG. 3, thereby amplifying the corresponding input to the input of the system 300. The channel signal is provided to provide Lout and Rout signals at outputs 316 and 318. The DVC system 302 can respond to the entire frequency range of each of the L and R signals, or only the selected frequency band of each signal. For example, as shown in FIG. 3, the high-pass filters 312 and 314 pass only the high-frequency portions of the individual L and R signals. To the DVC system 302, the latter responds only to the high frequency content of each signal.

Alternatively, a multi-channel system can be configured to allow a small number of selected channels to be individually processed by their respective DVC systems, thereby independently controlling the L and R signals. As shown in FIG. 4, dual channel system 400 employs two DVC systems 406 and 408 to provide L and R signals, such that the L and R signals applied to the inputs 402 and 404 are independently gain controlled. As shown, the L signal is applied to a high pass filter 410 and a low pass filter 412, and the R signal is applied to the high pass filter 414 and the low pass filter 416. In a dual channel system with high channel and low channel in Figure 4, a DVC system (406 and 408) can apply gain to the high channel by applying the output of each DVC system to the individual outputs of the high and low pass filters. L and R signals. In particular, the output of DVC system 406 is applied to control the gain of each of said amplifiers 418 and 420, while amplifiers 418 and 420 receive and amplify the high frequency outputs of high pass filters 410 and 412. Similarly, the output of DVC system 408 is applied to control the gain of each of 422 and 424 amplifiers, while amplifiers 422 and 424 receive and amplify the low frequency outputs of low pass filters 412 and 416. The outputs of the amplifiers 418 and 420 are added to the signal adder 426, thereby producing an output signal Lout at output 428; the outputs of the amplifiers 422 and 424 are added to the signal adder 430, thereby producing an output at output 432. Signal Rout.

In another embodiment, if the independent gain control of each L and R signal is desired in the multi-channel signal, the individual DVC systems can be used for each frequency band of the L and R signals. Further, by eliminating the multi-channel system, a high-pass filter can be used to eliminate the low-frequency portion, and the system does not respond to low frequencies, as shown in FIG.

Regarding the filter used in the multi-channel DVC system, the crossover frequency between each adjacent channel (which is a dual channel system of low pass and high pass channels) can be adjusted. It is also possible to keep the crossover frequency constant. One of the examples is based on a derivative of the Derived filter. For an explanation of the derived filter, refer to THAT Application Note 104, Audio Handbook (Section 5.2.4). An example of the implementation of the derivation filter, in which the second-stage Butterwortrh LPF and the resulting HPF are used in a cross, and the total is a unit, as shown in FIG. In another example, the intersection uses a fourth-stage Linkwitz-Riley filter, which is a unit, as shown in FIG. On the single channel volume control, the high pass filter controls the input to the RMS detector.

Multi-space processing protection (MPP)

TV manufacturers often incorporate virtual surround (like-surround) technologies (such as SRS Tru-Surround, Spatializer, etc.) into the two-channel TV sound output path. This two-channel TV sound may be transmitted to an external loudspeaker or to a loudspeaker placed in the TV stand. These virtual surround technologies create the illusion of surround sound by manipulating and enhancing the difference channel (L-R) currently in stereo broadcast. The listener is still aware of the complete center image (L+R), and often hears the difference channel (L-R), whether it is widening the studio or as a source outside the sound reinforcement position. This type of spatial enhancement is often carried out at the time of sound programming. This is especially true for TV ads that are enhanced to attract viewers' attention. When the sound program has two spatial enhancements in series (such as in production points and on TV sound processing), the sound quality may be greatly compromised. Pre-processed sounds are more prone to produce significant L-R energy relative to L+R energy. The second stage of the spatial enhancement process, the cascade stage tends to increase the amount of L-R energy. Recent studies have also shown that excessive L-R enhancement is one of the main causes of listener fatigue. Similarly, the volume may be greatly increased.

Thus, in accordance with one aspect of the present invention, an MPP system can be provided. In one embodiment, prior to the stereo enhancement technology of television, the MPP is a dual processing protection (DPP) system that is part of a television audio signal receiving and playback system. The MPP system, hereafter means a writaround surround signal processor. In the DPP system in the example, the processed audio signal minimizes the difference (LR) introduced by the production point; that is, minimizes the difference (LR) signal energy relative to the sum (L+R) signal. Level. This allows the TV's space enhancement technology to process the sound signal in a psychoacoustical way to please the listener. The DPP system has been proven to be effective in mitigating the harshness of dual spatial processing in the space of television to enhance the sound processing before the sound processing. In one embodiment, the DPP system is fully digital and can be implemented economically and efficiently using software (C, combined language, etc.) or digital hardware (HDL description language). It should be understood that the DPP system may be completely analogous or a mixture of analog and digital components.

In one embodiment, the DPP system reduces L-R enhancement relative to the corresponding L+R level. This embodiment reduces the effect of multi-dimensional two-channel spatial effect processing. The implementation of this system is shown at 800 in FIG. The left channel signal L and the right channel signal R are applied to inputs 802 and 804 of system 800, respectively. Signals L and R are applied to their matrix represented by two signal adders 806 and 808. The matrix formed by the signal adders 806 and 808 provides sum (L+R) and difference (L-R) signals.

In the (L+R) path, the signal is generally unaffected. The sum signal, which usually contains sound content, does not necessarily have to be localized. However, in an alternate embodiment, performing frequency contouring may enhance sound content, such as a conversation. As shown, the sum signal is first multiplied by the center constant at the signal multiplier 810 to provide a matrix of the two signal adders 812 and 814. The central constant allows the center image (L+R) to be adjusted as necessary to assist in the understanding of the dialogue. The L+R is added to the L-R signal to provide a left channel output signal L at output 816; and L+R is subtracted from L+R to provide a right channel output signal R at output 818.

In the embodiment illustrated in Figure 8, most of the processing occurs in the difference path. Comparing L+R with L-R determines the level of the L-R signal relative to L+R. Prior to comparison, the sum (SUM) and difference (DIF) signals can pass through high pass filters 820 and 822, respectively, such as when the loudspeaker frequency response does not include low frequencies. Right channel signal R. The L-R DIF signal can in turn pass through the multi-channel equalizer 824 to aggravate the most sensitive frequency of the ear, i.e., the intermediate frequency range, to complement the perceptible loudness level of the L-R signal. The equalizer 824 allows the detection of the difference channel level to be dependent on the frequency. For example, low frequency signals can be minimized when dealing with the limited bass response of inexpensive television loudspeakers. The high frequency can be minimized to limit the response of transient sound events. In general, the mid-range frequency, if the ear is most sensitive, can be controlled by the equalizer to control the difference level. Once the difference and sum signal levels are calculated, the DIF/SUM ratio can be determined.

Each of these signals will run through individual signal level detectors 828 and 830. Any of the above detectors can be used, such as an RMS level detector, although any type of level detector (such as each of the above detectors) can be used. At the same time, such processing can all be performed in the logarithmic domain and processed via log domain processing blocks 832 and 834 to increase efficiency.

The outputs of the blocks 832 and 834 are applied to the signal adder, wherein the processed SUM signal is subtracted from the processed DIF signal. Subtracting another signal from a signal in the logarithmic domain is as if a signal is provided, which is the ratio of the ratio of the SUM signal to the DIF signal in the linear domain. Once the level of the L+R and LR signals is calculated (where the LR signal level may have been equalized before the level detection to improve the mid-range frequency), the two signal levels are compared by comparator 838 for Preset threshold value 840. The ratio between the two signals ((L-R)/(L+R)) is compared to the threshold ratio of comparator 838 to determine the proposed L-R gain adjustment. Limiter stage 842 can be used to limit the amount and direction of gain applied to the L-R signal. The illustrated embodiment limits the gain to 0 dB and thus only allows attenuation of the L-R signal, although some applications may want to amplify the L-R signal. An averaging stage 844 averages the output of the limiter stage 842 with a relatively long time constant, thereby preventing the DPP system from tracking short temporary sound events. After being converted back to the linear domain by linear domain block 846, the level of the L-R signal is thus adjusted by the signal multiplier 848 to achieve the target ratio.

That is, if there is no multi-stage spatial processing, the target ratio (L-R)/(L+R) can be set low, so that the program dialogue can be improved.

Another method and system for dual processing protection is to "pre-announce" its pre-processing of the L-R signal and supplement it with the advance notice. For example, if SRS Tru-Surround is known for L-R, the signal can therefore be sufficient to remove the L-R enhancement. Alternatively, the signal energy can be monitored at any time to infer its pre-processing of the L-R signal. After this interpretation, the inference can be made up by removing any L-R enhancements. Pre-processing may change the frequency response of the difference channel (and the channel) and the L-R/L+R ratio. The inverting filter of the pre-processor can be applied to each path while the existing L-R/L+R ratio adjustment is still in use.

Further, when the DPP system shown in FIG. 8 is used as a feedforward system, the sensing of the DIF signal precedes the variable gain control amplifier 848, a feedback system, and the difference signal is performed after the variable gain control amplifier. Precise detection.

Combine DVC with DPP

Since each DVC and MPP provide enhanced listening experience, the two can be combined to combine the advantages of both parties. There are many ways to combine DVC and DPP blocks. A useful topology example, first place DPP block 902, and then design DVC block 904 in cascade, as shown in FIG. In this embodiment, the L and R signals are applied to inputs 906 and 908 of DPP block 902. DPP block 902, the L' and R' signals of outputs 910 and 912, are applied to two inputs 914 and 916 of DVC block 904. Outputs 918 and 920 of the DVC block provide individual output signals Lo and Ro. This cascade design allows the DPP block to remove the difference (L-R) signal enhancement first, thereby maintaining the perceived constant level of the stereo program with the DVC block without the need for ambient energy.

In another topology example, the DPP block 1004 is placed in the feedback path of the DVC block 1002, as shown in FIG. The L and R inputs are applied to inputs 1006 and 1008, respectively. The two signals are applied to the matrix (represented by signal adders 1010 and 1012), thereby generating SUM (L+R) signals and DIF (L-R) signals. Outputs 1014 and 1016 of the DVC block 1002 provide output signals Lo and Ro. The two outputs 1014 and 1016 provide a feedback signal of the feedback path. In particular, the Lo and Ro signals are applied to the matrix, as shown by adders 1018 and 1020, such that Lo+Ro forms one input to DPP block 1004 and Lo-Ro forms another input of DPP block 1004. The output of DPP block 1004 represents the corrected gain, which is applied to the DIF signal by signal multiplier 1022; in the form of a multiplier, it can be a variable gain control amplifier. It should be understood that although the second embodiment of combining DVC and DPP blocks is illustrated in Figures 9 and 10, other combinations are still possible.

Therefore, the presently disclosed embodiments are capable of improving the performance of voice signal reproduction and reducing the effects of undesired volume changes on sound program planning.

The components, steps, features, benefits, and advantages discussed are merely illustrative. None of them or their related discussions intends to limit the scope of protection in any way. Many other implementations are also available. In addition, the presently disclosed embodiments may have fewer components, steps, features, advantages and advantages than those indicated herein. These also include different configurations and/or orders of components and/or steps in the embodiments.

Unless otherwise mentioned, all measurements, values, ratings, rankings, sizes, sizes, and other specifications expressed in this specification (including the scope of patents requested later) are generally (not entirely true). They are intended to have a reasonable range, consistent with their relevant functions, and consistent with their technical conventions.

All articles, patents, patent applications, and other publications cited herein are hereby incorporated by reference.

Where the term "intention" is used in the context of a patent claim, the intent and the explanation are intended to cover the corresponding structures and materials described, and their equivalents. In the same way, the term "step" is used in the context of a patent claim, and covers the corresponding actions described and their equivalents. If such terms are not used, it is intended that the scope of the invention is not intended to be

The reference to or to the present invention, whether or not it is referred to in the claims of the patent claims, is not intended to be construed as a limitation to any of the components, steps, features, advantages, advantages, or equivalents.

The scope of protection is limited only by the following patent scope requests. The intent and clarification of the scope of the scope is that the breadth of the scope is consistent with the normal meaning of the language used in the patent scope request and is equivalent in all structural and functional terms. Things.

100. . . DVC system

102. . . Input

104. . . Input

110. . . RMS level detector

112. . . Logarithmic conversion block

114. . . Signal average AVG square

116. . . Target signal generator

118. . . Rising valve limit signal device

120. . . Gate valve limit signal device

122. . . Ratio signal device

124. . . Silent locking device

126. . . Signal adder

128. . . Comparators

130. . . Comparators

132. . . Signal adder

134. . . Signal adder

136. . . Signal multiplier

138. . . Rise/release selection block

140. . . Signal converter

142. . . Signal multiplier

144. . . Signal multiplier

200. . . operation

202. . . Mute off state

208. . . Mute on state

212. . . Mute lock status

300. . . Single channel system

302. . . DVC system

304. . . Input

306. . . Input

308. . . Amplifier

310. . . Amplifier

312. . . High pass filter

314. . . High pass filter

316. . . Output

318. . . Output

400. . . Channel system

402. . . Input

404. . . Input

406. . . DVC system

408. . . DVC system

410. . . High pass filter

412. . . Low flux waver

414. . . High flux wave

416. . . Low flux waver

418. . . Amplifier

420. . . Amplifier

422. . . Amplifier

424. . . Amplifier

426. . . Signal adder

428. . . Output

430. . . Signal adder

432. . . Output

800. . . system

802. . . Input

804. . . Input

806. . . Signal adder

808. . . Signal adder

810. . . Signal multiplier

812. . . Signal adder

814. . . Signal adder

816. . . Output

818. . . Output

820. . . High pass filter

822. . . High pass filter

824. . . Multi-band equalizer

828. . . Signal level detector

830. . . Signal level detector

832. . . Logarithmic domain processing block

834. . . Logarithmic domain processing block

838. . . Comparators

840. . . Preset threshold

842. . . Limiter stage

844. . . Average stage

846. . . Linear domain block

848. . . Signal multiplier

902. . . DPP box

904. . . DVC box

906. . . Input

908. . . Input

910. . . Output

912. . . Output

914. . . Input

916. . . Input

918. . . Output

920. . . Output

1002. . . DVC box

1004. . . DPP box

1006. . . Input

1008. . . Input

1010. . . Signal adder

1012. . . Signal adder

1014. . . Output

1016. . . Output

1018. . . Signal adder

1020. . . Signal adder

1022. . . Signal multiplier

The disclosure of the drawings is intended to illustrate the embodiments and not all embodiments. Other embodiments may be used alternatively or alternatively. In order to save space or for more effective explanation, details that are clearly known or unnecessary may also be omitted. Conversely, in some implementations, the details disclosed may not be present in practice. The same numbering in different figures means the same or similar components or steps.

The aspects disclosed herein are more fully understood as the following description is read in conjunction with the accompanying drawings, which are considered as illustrative and not limiting. The drawings are not necessarily drawn to scale, but rather to emphasize the principles disclosed. In the drawings:

1 is a simplified block diagram of one embodiment of a dynamic volume control system;

2 is a state diagram illustrating an implementation of a program transition detecting its operation;

3 is a simplified block diagram of one embodiment of a single frequency band of a dynamic volume control system;

4 is a simplified block diagram of one embodiment of a multi-band dynamic volume control system;

Figure 5 to Figure 7 show the frequency response of the multi-band dynamic volume control system;

Figure 8 is a simplified block diagram of one embodiment of a dual processing protection system;

9 is a simplified block diagram of one embodiment of a combined system configuration, including a dynamic volume control system and a dual processing protection system;

10 is a simplified block diagram of a second embodiment of a combined system configuration, including a dynamic volume control system and a dual processing protection system.

Claims (7)

  1. A system for dynamically controlling a perceived volume of a stereo program including one of left and right channel signals, comprising: a dynamic volume control system configured and configured to maintain a perceptual constant volume level of the stereo program The dynamic volume control system includes a signal level sensor for sensing the average level of the left and right channel signal levels; and a signal average block configured and configured to provide a rise Time and a release time, and signal compression is generated according to a compression ratio, wherein the signal average block is further configured to, for a gain value to be applied to the left and right channel signals, relative to the rise time and the release time, Calculating and providing an average value (AVG) of a reference (REF) signal as an output signal, wherein the reference (REF) signal represents a volume level of the input signal relative to the threshold value to be listened to; and an excess space processing a protection processor configured to control a difference signal bit generated by a function of subtracting the right channel signal (LR) from the left channel signal One of the signal and the signal level generated by the function of the right channel signal plus the left channel signal; wherein the excess space processing protection processor performs the sound signal processing, and is controlled to be relative to the sum ( The difference (LR) signal of the L+R) signal.
  2. The system of claim 1, wherein the excess spatial processing protection processor is in series with the dynamic volume controller.
  3. The system of claim 2, wherein the excess space processing protection processor is cascaded to the front of the dynamic volume controller, and thus first controls its difference (LR) signal enhancement relative to the sum (L+R) signal, thereby The perceptual steady level of the stereo program is maintained by the dynamic volume control system without abnormal peripheral difference energy.
  4. The system of claim 1, wherein the excess spatial processing protection processor resides in a feedback path of the dynamic volume control system.
  5. The system of claim 1, wherein the dynamic volume control system responds to one or more of the following signals: a target level signal indicating the desired volume level of the sum of the left and right channel signal levels, wherein a difference The signal indicates the difference between the sensed average level and the target level signal; a rising threshold signal indicates that the dB number of the difference signal must be above the set point before the rise time is increased by N times; a release threshold signal , indicating that the dB number of the difference signal must be below the set point before the release time is increased by M times; a rising ratio threshold signal indicates that the difference signal can be before the dynamic volume control system starts to attenuate the left and right channel signals. The absolute amount of dB exceeding a set point; a release ratio threshold signal indicates that the differential signal can be lower than a set point of the absolute amount before the dynamic volume control system begins to add gain to the left and right channel signals. a proportional signal, which can be used to adjust the sensing level according to a desired compression ratio Average level.
  6. The system of claim 5, wherein the values of N and M are each 10.
  7. A method for dynamically controlling a perceived volume of a stereo program including one of left and right channel signals, the method comprising: dynamically controlling a volume level of a stereo program, thereby maintaining the volume level at a constant volume level of perception Dynamically controlling the volume level includes sensing the average level of the left and right channel signal levels; and providing a rise time and a release time, generating a signal compression according to a compression ratio, and calculating and providing a An average value (AVG) of the reference (REF) signal is used as an output signal, wherein the reference (REF) signal represents a volume level of the input signal relative to the threshold value to be listened to; and the level of the difference signal is controlled, And the signal generated by subtracting the function of the right channel signal (LR) from the left channel signal, and the signal and the signal generated by the function of adding the right channel signal to the left channel signal; wherein the control includes processing The voice signal is thus controlled to enhance its difference (LR) signal relative to the sum signal (L+R).
TW098138834A 2008-11-14 2009-11-16 Dynamic volume control and multi-spatial processing protection TWI491277B (en)

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Citations (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4866774A (en) * 1988-11-02 1989-09-12 Hughes Aircraft Company Stero enhancement and directivity servo
US5127059A (en) * 1990-01-18 1992-06-30 Gibson Guitar Corp. Audio amplifiers
TW200623024A (en) * 2004-12-21 2006-07-01 Dolby Lab Licensing Corp Calculating and adjusting the perceived loudness and/or the perceived spectral balance of an audio signal
US20060256980A1 (en) * 2005-05-11 2006-11-16 Pritchard Jason C Method and apparatus for dynamically controlling threshold of onset of audio dynamics processing
TW200746049A (en) * 2006-04-04 2007-12-16 Dolby Lab Licensing Corp Calculating and adjusting the perceived loudness and/or the perceived spectral balance of an audio signal
US20080253586A1 (en) * 2007-04-16 2008-10-16 Jeff Wei Systems and methods for controlling audio loudness

Patent Citations (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4866774A (en) * 1988-11-02 1989-09-12 Hughes Aircraft Company Stero enhancement and directivity servo
US5127059A (en) * 1990-01-18 1992-06-30 Gibson Guitar Corp. Audio amplifiers
TW200623024A (en) * 2004-12-21 2006-07-01 Dolby Lab Licensing Corp Calculating and adjusting the perceived loudness and/or the perceived spectral balance of an audio signal
US20060256980A1 (en) * 2005-05-11 2006-11-16 Pritchard Jason C Method and apparatus for dynamically controlling threshold of onset of audio dynamics processing
TW200746049A (en) * 2006-04-04 2007-12-16 Dolby Lab Licensing Corp Calculating and adjusting the perceived loudness and/or the perceived spectral balance of an audio signal
US20080253586A1 (en) * 2007-04-16 2008-10-16 Jeff Wei Systems and methods for controlling audio loudness

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