Output Power Limiter in an Audio Amplifier
An output power limiter system (PLS) for audio amplifiers may be designed as a feedback control system for protection of the load and/or quality of the audio experience. The PLS may use comparator to sense the output current, compare it to a specified threshold, and assert a signal when the output current reaches and exceeds the specified threshold. The output signal from the comparator may enable a counter that is clocked with a high frequency clock to begin counting to measure the pulse-width of the comparator output. The output of the counter may be averaged through a fast attack and slow release infinite impulse response (IIR) filter having programmable settings to generate a rate of attenuation or rate of release that adjusts a gain correction in terms of decibels (dB). The output of the IIR filter may then be used for attenuating the output current.
This application claims benefit of priority of U.S. provisional application Ser. No. 61/543,998 titled “Circuits and Methods Used in Audio Signal Processing”, filed Oct. 6, 2011, which is hereby incorporated by reference in its entirety as though fully and completely set forth herein.
FIELD OF THE INVENTIONThe present invention relates generally to signal processing and mixed signal circuits, and more particularly to limiting output power in an audio amplifier.
DESCRIPTION OF THE RELATED ARTSignal processing represents a combined application of electrical/computer engineering and mathematical principles, primarily directed to the analysis of and operation on either discrete or continuous time signals. Signals of interest can include sound, images, time-varying measurement values and sensor data, for example biological data such as electrocardiograms, control system signals, telecommunication transmission signals such as radio signals, and many others. Signals are typically analog and/or digital electrical representations of time-varying or spatial-varying physical quantities. Types of signal processing include analog, discrete time, and digital.
Analog signal processing is performed on signals that have not been digitized, for example signals that are used in classical radio, telephone, radar, and television systems. Analog signal processing typically makes use of linear electronic circuits such as passive filters, active filters, additive mixers, integrators and various types of delay lines, as well as non-linear circuits such as frequency mixers and voltage-controlled amplifiers, voltage-controlled filters, voltage-controlled oscillators and phase-locked loops. Discrete time signal processing is performed on sampled signals that are defined at discrete points in time, and as such are quantized in time, but not in magnitude. Analog discrete-time signal processing is based on electronic devices such as sample and hold circuits, analog time-division multiplexers, analog delay lines and analog feedback shift registers, and may be considered a predecessor of digital signal processing.
Digital signal processing involves the processing of digitized discrete-time sampled signals. Processing is typically performed by general-purpose computers or digital circuits such as application specific integrated circuits (ASICs), field-programmable gate arrays, or specialized digital signal processors (DSPs). Digital signal processing mostly includes performing arithmetic operations such as fixed-point and floating-point operations, real-valued and complex-valued operations, multiplication and addition. Many of these operations are implemented through the use of circular buffers and look-up tables. Examples of digital signal processing algorithms include Fast Fourier transforms (FFT), finite impulse response (FIR) filters, infinite impulse response (IIR) filters, and adaptive filters such as the Wiener and Kalman filters.
Audio signal processing, sometimes referred to as audio processing, is the processing of electrical signals that correspond to auditory signals, or sound. Since audio signals may be electronically represented in either digital or analog format, audio signal processing may also take place in either the analog or digital domain. In analog audio signal processing, operations are performed directly on the electrical signals corresponding to the audio signals, while digital signal processing consists mostly of mathematical operations performed on digital representations of the electrical signals that correspond to respective audio signals. Typically, the digital representation of audio signals expresses the pressure waveform that characterizes the audio signal as a sequence of binary numbers. This permits signal processing using digital circuits such as microprocessors and computers, and while analog to digital conversion can be prone to loss, most modern audio systems use the digital approach because digital signal processing techniques are overall more powerful and efficient than signal processing in the analog domain.
Overall, since audio signals first need to be converted to electrical signals, digital audio processing systems include both analog and digital components in a full processing path that begins with the pressure waveforms that physically define the audio signal and ends with the digital representation of the corresponding electrical signals derived therefrom. Some of the most common components typically used in audio processing systems include pulse-width modulators, power limiters, start-up circuits, power regulators, comparators, amplifiers, oscillators, among others. The quality and operating precision of these components directly impacts the quality of audio signal processing systems, as designers have to continually overcome numerous difficult design challenges to meet required specifications and quality standards.
SUMMARYVarious embodiments of an amplifier feature the capability of limiting output power in the amplifier. The amplifier may include two or more pulse-width modulators (PWM) controlling respective sets of switches, and for large input signals the output load may become susceptible to potential damage. Therefore, it is desirable to attenuate the output of the amplifier in a controlled manner that protects the load without interrupting or compromising the quality of the audio signal in any way. In one set of embodiments, output power in the amplifier may be limited through a pseudo-power limiter, which may be operated to reduce the width of the PWM pulse to the gate of the high-side FET in a push-pull bridge configuration, to effectively limit the amount of current that passes through the load. The output power is defined as the product of the output voltage and output current, P=V*I.
The power limiter may include a mixed-signal system that uses a simple analog-to-digital converter (ADC) to sense the output current level of the corresponding audio output signal and compare this current level to a specified threshold current level. The power limiter may also include a digital circuit that samples the error output, filters out the error, and attenuates the output audio through direct gain scaling of the PWM output pulse. In one set of embodiments, the ADC with the built-in error detection threshold may be as simple as an analog comparator that asserts a signal when its input is greater than the error threshold. Subsequently, the output signal from the comparator may be used to enable a counter clocked with a high frequency clock to begin counting in order to measure the pulse width, and therefore essentially digitize the error (i.e. it obtains a numeric/digital value corresponding to the error). The digital filter may be a first, second or higher order infinite impulse response (IIR) filter to average the error. In some embodiments, the circuit used for attenuating the output audio may be a fast attack and slow release IIR filter with programmable settings, to generate a rate of attenuation or rate of release that adjusts a gain correction in terms of decibels (dB). The gain correction circuit, which attenuates the width of the PWM output pulse, may be a circuit similar to a volume adjustment circuit used in audio systems.
In one set of embodiments, the audio amplifier may be an open-loop amplifier or a closed-loop amplifier in terms of accepted industry audio performance specifications such as signal to noise ratio (SNR), total harmonic distortion plus noise (THD+N), cross-talk, etc. The limiter system may be designed as a feedback control system for either protection of the load or quality of the audio experience.
A better understanding of embodiments of the present invention may be obtained when the following Detailed Description is considered in conjunction with the following drawings, in which:
While the invention is susceptible to various modifications and alternative forms, specific embodiments thereof are shown by way of example in the drawings and are herein described in detail. It should be understood, however, that the drawings and detailed description thereto are not intended to limit the invention to the particular form disclosed, but on the contrary, the intention is to cover all modifications, equivalents and alternatives falling within the spirit and scope of the present invention as defined by the appended claims.
DETAILED DESCRIPTIONThe Digital PWM block 204 includes a PWM Controller 212, two PWM driver blocks labeled PWM0 (232) and PWM1 (234), and may contain other logic as well. The PWM Controller 212 calculates edge locations of the pulse train to be generated, and produces two (M−1)-bit outputs. In particular, each of the two (M−1)-bit outputs may represent respective edges of pulses to be generated. The two (M−1)-bit outputs are used by individual pulse width modulators PWM0 and PWM1 to produce the final differential PWM outputs PWM_OUTP and PWM_OUTN. In general, PWM block 204 may comprise a small signal-processing block that operates on the M-bit input data and separates the M-bit input data into two individual streams of M−1 bits each. These (M−1)-bit streams may be independent, or, more specifically, they may have some correlation to each other, while the actual data may differ on an instantaneous pulse-by-pulse basis.
Block 222 comprises logic for handling dead time, as well as a MOSFET Power Output Stage and gate drivers for controlling the MOSFET Power Output Stage. The MOSFET Power Output Stage portion of block 222 may include high power switches, preferably MOSFETs (Metal Oxide Semiconductor Field Effect Transistors). The high power switches generate a high-power (amplified) replica of the received pulse train. The MOSFET Power Output Stage portion provides the amplified pulse train to low pass filter 224. As shown, the Output Stage provides a differential pair of output signals, referred to as OUTP and OUTN, which provide two differential pulses per PWM period. The low pass filter 224 performs a low pass filter operation on the differential pulses and provides the two outputs, referred to as OUTP_FILT and OUTN_FILT, to a load, e.g., to a loudspeaker 124.
Adaptive Output Power LimiterFor large input signals, the output load, for example loudspeaker 124 may become susceptible to potential damage. Therefore, it may be desirable to attenuate the output of the amplifier in a controlled manner to protect the load, but without interrupting or compromising the quality of the audio signal. In general, power limiting is used in audio applications to protect the loudspeakers from failures. However, it might be equally desirable to limit output power in a controlled manner in other applications, for example at the output of a switching power supply, or other similar circuits. In audio applications, the aforementioned failures may occur when excessive power is driven into the loudspeaker (e.g. speaker 124) for an extensive amount of time. Traditionally, true power limiting has been implemented by restricting the power delivered to the loudspeaker by measuring the actual current and actual voltage at the output of the amplifier. In one set of embodiments, an adaptive audio power limiter provides a very low cost alternative to present day power limiters.
Output power is determined by equation P=V*I, where V is the output voltage and I is the output current. It follows from the above equation that output power may be limited if either the voltage or the current is used as a feedback variable to limit the total power in the system. In one set of embodiments, a power limiting circuit may be used to measure the output current, generate a control output based on the measured output current, and use the control output to adaptively limit the output power. For example, the control output may be converted to a single signal, e.g. a digital signal, which may be processed by a digital circuit to adaptively limit the power to the loudspeaker without fear of immediate shutdown. Various embodiments of the adaptive power limiter may be used in various applications, for example in an audio subsystem.
Traditionally, power stages associated with audio amplifiers have included a fault detection circuit operated to shut down the PWM signals OUTP and OUTN to the H-Bridge 442 based on an over-current event. As a result of this action, the loudspeakers are protected from excessive power. Typically, a fixed threshold is set for the current, above which the current would cause the device to shut down. One example of such a prior art current-shutdown circuit topology is shown in
The waveforms of interest corresponding to the power limiting implementation, shown in
As shown in
In one set of embodiments, each of the thresholds may be programmable. For example, the outer and inner comparators may be programmed with absolute fixed thresholds, as illustrated by diagram 706 in
The topology of a further improved and even more flexible power limiting circuit according to another embodiment is illustrated in
The digital controller (934/944), labeled “pulse-width detector” may be a digital signal processor (DSP) or a microprocessor, or it may be implemented as glue logic, or a combination of hardware and software, and may operate to measure the width of the signal provided by blocks 812/814. The resulting signals from pulse-width detectors 934/944 are then filtered by respective filters 932 and 942, which are coupled to processing elements 934/944 to receive the respective outputs therefrom. In the embodiment shown in
As shown in
As previously mentioned, to obtain the effect illustrated in
In one set of embodiments, the filters 932 and 942 may also serve as the scaler control. That is, filters 932/942 and scaler calculators 936/946 may respectively be implemented as a single processing block to provide the multiplier value with which the output from audio signal processor 202 is multiplied to scale the PWM pulses. The type of filter used in this instance may be a fast attack and slow release filter. The power limiter may attack fast to ensure that the fault shutdown event does not occur. The undesired audible effects will be less audible, since the actual audio may be loud when the protection afforded by the power limiter circuit is required. The release may be slow to ensure that the volume change is less noticeable or audible since the actual audio will be less loud because there will be less current required to support the actual audio. The architecture of this fast attach and slow release filter may be implemented with a single, reprogrammable IIR filter in which the value of the integrator 952 is shifted by a factor of 2, either smaller or larger, depending on the instantaneous change of modes, which include attack to release and release to attack. This is show in the embodiment of the IIR filter in
Although the embodiments above have been described in considerable detail, numerous variations and modifications will become apparent to those skilled in the art once the above disclosure is fully appreciated. It is intended that the following claims be interpreted to embrace all such variations and modifications.
Claims
1. A signal processing system comprising:
- signal processing circuitry configured to receive an input signal, generate a driver signal representative of the input signal, and use the driver signal to generate an output signal; and
- power limiting circuitry configured to:
- assert a first signal in response to an output current reaching a first threshold, wherein the output current is produced by the output signal;
- derive a control value from the first signal, wherein the control value corresponds to a time duration for which the first signal is asserted;
- attenuate the output signal by scaling the driver signal according to the control value.
2. The signal processing system of claim 1, wherein the signal processing circuitry is an amplifier circuit, wherein the input signal is an audio signal, and wherein the output signal is an amplified version of the audio signal.
3. The signal processing system of claim 1, further comprising a speaker coupled as a load to the signal processing circuitry, wherein the output current is conducted by the speaker.
4. The signal processing system of claim 1, wherein the power limiting circuitry comprises:
- a comparator having an output configured to provide the first signal responsive to the comparator comparing the output current to the first threshold;
- a digital circuit configured to generate a numeric value based on the first signal, wherein the numeric value corresponds to the time duration for which the first signal is asserted; and
- a filter circuit configured to generate the control value from the numeric value.
5. The signal processing system of claim 4, wherein the digital circuit is a counter that is clocked with a clock signal having a specified frequency higher than a frequency of the driver signal;
- wherein the counter is configured to: count when the first signal is asserted; stop counting when the first signal is deasserted; and provide a result of the count as the numeric value.
6. The signal processing system of claim 4, wherein the filter circuit is an infinite impulse response (IIR) filter configured to average the numeric value, and provide the control value based on the averaged numeric value.
7. The signal processing system of claim 6, wherein the IIR filter is a fast attack and slow release IIR filter.
8. The signal processing system of claim 7, wherein the IIR filter comprises programmable settings for generating a rate of release.
9. The signal processing system of claim 1, wherein the power limiting circuitry is further configured to:
- deassert the driver signal in response to the output current reaching a second threshold.
10. The signal processing system of claim 9, wherein the second threshold is higher than the first threshold.
11. The signal processing system of claim 1, wherein the driver signal is a pulse-width modulated (PWM) signal;
- wherein in scaling the driver signal according to the control value, the power limiting circuit is configured to adjust a pulse-width of the PWM signal.
12. A power limiting circuit for limiting output power associated with an output current resulting from an output signal generated according to a driver signal, which is derived from an input signal, the power limiting circuit comprising:
- a comparator configured to assert a comparator output responsive to the output current exceeding a first threshold value; and
- a scaler circuit configured to: derive a scaler value from the comparator output; and attenuate the output signal by multiplying the scaler value with the input signal.
13. The power limiting circuit of claim 12, wherein the scaler circuit is configured to generate the scaler value based on a time duration for which the comparator output is asserted.
14. The power limiting circuit of claim 12, wherein the scaler comprises:
- a digital circuit configured to generate a numeric value corresponding to a time duration for which the comparator output is asserted; and
- a filter circuit configured to generate the scaler value from the numeric value.
15. The power limiting circuit of claim 14, wherein the digital circuit is a counter that is clocked with a clock signal having a specified frequency higher than a frequency of the driver signal;
- wherein the counter is configured to: count when the comparator output is asserted; stop counting when the comparator output is deasserted; and provide a result of the count as the numeric value.
16. The power limiting circuit of claim 14, wherein the filter circuit is an infinite impulse response (IIR) filter configured to average the numeric value, and provide the scaler value based on the averaged numeric value.
17. The power limiting circuit of claim 16, wherein the IIR filter is a fast attack and slow release IIR filter.
18. The power limiting circuit of claim 17, wherein the IIR filter comprises programmable settings for generating a rate of release.
19. The power limiting circuit of claim 14, further comprising:
- a second comparator configured to deassert the driver signal in response to the output current reaching a second threshold.
20. The power limiting circuit of claim 19, wherein the second threshold is higher than the first threshold.
21. A method for limiting output power in an audio amplifier, the method comprising:
- generating an output signal based on an input signal;
- the output signal resulting in an output current;
- asserting a control signal when the output current is over a specified first threshold;
- generating a numeric value corresponding to a time duration for which the control signal is asserted;
- averaging the numeric value;
- scaling the averaged numeric value; and
- attenuating the output signal by multiplying the scaled averaged numeric value with the input signal.
22. The method of claim 21, wherein said generating the numeric value comprises:
- counting cycles of a clock signal while the control signal is asserted; and
- providing a number of counted clock cycles as the numeric value.
23. The method of claim 21, wherein said averaging the numeric value comprises filtering the numeric value using an infinite impulse response (IIR) filter.
24. The method of claim 21, wherein said scaling the averaged numeric value comprises performing one of:
- a successive approximation register algorithm using the averaged numeric value as input; and
- a linear reduction using the averaged numeric value as input.
25. The method of claim 21, wherein said generating the output signal based on the input signal comprises:
- generating a pulse train representative of the input signal; and
- generating the output signal by driving a power stage with the pulse train;
- wherein the method further comprises adjusting the pulse train responsive to said multiplying the scaled averaged numeric value with the input signal.
Type: Application
Filed: Aug 20, 2012
Publication Date: Apr 11, 2013
Inventors: Douglas E. Heineman (Lakeway, TX), Michael S. Pate (Austin, TX)
Application Number: 13/589,303