CN115426590A - Digital audio anti-jitter and delay transmission prevention audio amplifier - Google Patents
Digital audio anti-jitter and delay transmission prevention audio amplifier Download PDFInfo
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Abstract
The invention relates to a digital audio anti-jitter and delay transmission prevention audio amplifier, which is suitable for application scenes such as live broadcast, musical instrument playing and the like which need real-time feedback and have extremely high delay requirements. The method comprises the steps that an overclocking controller and an anti-shake processing module are arranged, the overclocking controller obtains a delay change curve, time integration is carried out, an integration result D is input into an adjusting model, and overclocking frequencies f corresponding to different D values are calculated in the adjusting model; the overclocking controller obtains overclocking frequency f and sends to DSP audio frequency processing module, and DSP audio frequency processing module works according to overclocking frequency f to realize DSP audio frequency processing module frequency conversion work, reduce delay transmission. The anti-shake processing module is used for generating anti-shake reverse phase signals and outputting the anti-shake reverse phase signals to the output interface, and the anti-shake reverse phase signals are superposed with signals output by the digital-to-analog conversion amplification module, so that an anti-shake effect is achieved.
Description
Technical Field
The present invention relates to audio amplifiers, and more particularly, to an anti-jitter and delay-tolerant digital audio amplifier.
Background
Digital Signal Processing (DSP) is an emerging subject that relates to many subjects and is widely used in many fields. With the rapid development of computer and information technology since the 60 s of the 20 th century, digital signal processing technology has come to work and has been rapidly developed. Digital signal processing has been used very widely in the field of communications over the last two decades. The digital signal processing is to use a computer or a special processing device to acquire, transform, filter, estimate, enhance, compress, identify and the like the signals in a digital form so as to obtain a signal form meeting the needs of people.
The current literature and literature development is changing day by day, and industries such as digital musical instruments and live broadcasting develop rapidly, so that a large amount of DSP audio processing equipment is needed in the field, and meanwhile, the DSP is integrated in an audio amplifier quite often. At present, a DSP module is generally in a fixed-frequency working mode, and when a large data volume is encountered, delay instability can be caused, and large delay and jitter are generated. In order to overcome the situation, the working dominant frequency of the DSP equipment needs to be continuously improved, but the chip with the high dominant frequency is generally higher in cost, and meanwhile, the chip is overheated and the service life is reduced due to the fact that the chip generates larger heat when the chip works with the high dominant frequency.
Disclosure of Invention
In view of the above, to solve the above problems, an audio amplifier for digital audio anti-jitter and anti-delay transmission is provided, which includes an input interface, an output interface, a DSP audio processing module, a digital-to-analog conversion amplifying module, a delay calculating module, a delay feedback module, an over-frequency controller, and an anti-jitter processing module;
the input interface is used for connecting a sound source, acquiring original audio data, and sending the original audio data to the DSP audio processing module;
the DSP audio processing module processes audio, and the processing comprises gain, distortion, EQ adjustment and reverberation increase;
the audio data output by the DSP is sent to a digital-to-analog conversion amplifying module, the digital-to-analog conversion amplifying module performs digital-to-analog conversion on the audio data to obtain an analog audio signal, and the analog audio signal is output to an output interface after being amplified;
the delay calculation module acquires the delay t between the input interface and the output interface and sends the delay to the delay feedback module; the delay feedback module sends the delay t to the over-frequency controller; the over-frequency controller is connected with the DSP audio processing module and the anti-shake processing module;
the over-frequency control module is used for controlling the over-frequency performance of the DSP processing chip, the anti-shake processing module is used for generating anti-shake reverse phase signals and outputting the anti-shake reverse phase signals to the output interface, and the anti-shake reverse phase signals are superposed with signals output by the digital-analog conversion amplification module, so that an anti-shake effect is achieved.
The input interface is provided with a marking module, the marking module is used for superposing a feature code on the original audio data acquired by the input interface, the feature code comprises a timestamp, and the feature code corresponds to a feature frequency after being subjected to digital-to-analog conversion; the feature code can be identified and analyzed by the delay calculation module, so that a time stamp of the feature code is obtained;
the feature codes are sent to the DSP audio processing module and then output to the digital-to-analog conversion and amplification module, and the feature codes are converted into analog signals and sent to an output interface;
the time delay calculation module acquires the characteristic frequency corresponding to the characteristic code from the output interface and calculates the time of the characteristic frequency, so that the time delay delta t of the characteristic code which is output to the digital-to-analog conversion amplification module and then transmitted to the output interface after passing through the DSP audio processing module is calculated.
The delay feedback module acquires the delay delta t obtained by the delay calculation module in real time; generating a time-delay delta t change curve along with time, namely a time-delay change curve, and sending the time-delay change curve to the overclocking controller;
the overclocking controller obtains a time delay change curve and carries out time integration, wherein the time range of the integration is t from the current moment 1 Integrating to the current time; inputting the integration result D into an adjusting model, and calculating the overclocking frequency f corresponding to different D values in the adjusting model; the overclocking controller obtains overclocking frequency f and sends to DSP audio frequency processing module, and DSP audio frequency processing module is according to overclocking frequency f work to realize DSP audio frequency processing module frequency conversion work, reduce delay transmission.
The modeling method of the adjusting model comprises the following steps:
inputting the same section of audio signal into a DSP audio processing module, and performing single processing and superposition mixing processing by using four processing modes of gain, distortion, EQ adjustment and reverberation increase in the DSP audio processing module under basic working frequency, wherein the four processing modes are 15 combinations; the time delay delta t under 15 groups of different processing conditions is obtained by using a time delay calculation module 0 (ii) a Adjusting the working frequency of the DSP audio processing module to enable the DSP audio processing module to work in different over-frequency f states, and then obtaining different frequencies by using the delay calculation moduleThe time delay delta t of the lower 15 groups under different treatment conditions is obtained to obtain 15 groups of f-delta t curves;
let the delay time of each combination when the DSP audio processing module is working at the highest over-frequency be Deltat 1 (ii) a Calculating coefficient M = ([ Delta ] t) 0 -△t 1 )/△t 1 (ii) a The M value represents the sensitivity of the delay of the DSP audio processing module to the working frequency in different working modes, the larger the M value is, the more sensitive is, and the more obvious the delay reduction realized by increasing the frequency is;
setting a threshold D in the tuning model 0 When D > D 0 Then, the frequency Δ f = (k/M) · f is increased 0 (ii) a When D is less than D 0 When the frequency is reduced, Δ f = (k/M) · f 0 (ii) a Wherein f is 0 To adjust the step size.
The anti-shake processing module obtains analog audio data from the digital-to-analog conversion amplifying module and uses the audio loudness A 0 Dividing the analog audio data for threshold value when the actual loudness A of the analog audio data is larger than A 0 At the same time, the loudness exceeds A 0 When the actual loudness A of the analog audio data is less than A 0 Then, the corresponding part is copied, and the loudness after copying is adjusted to A 0 -A；
The delay feedback module acquires the delay delta t obtained by the delay calculation module in real time; generating a time-delay delta t change curve, namely a time-delay change curve, and sending the time-delay change curve to the anti-shake processing module;
the anti-shake processing module calculates t from the current moment to the front 1 Outputting the inverted audio and the copied audio generated by the anti-shake processing module to an output interface for superposition when the variance of delta t in the time period of the current moment exceeds a threshold value; and when the variance is smaller than the threshold value, outputting the reverse audio generated by the anti-shake processing module and the copied audio to an output interface for superposition.
Furthermore, a phase processing module is arranged behind the DSP audio processing module; the phase processing module is connected between the DSP audio processing module and the digital-to-analog conversion amplification module and is used for performing phase processing on the audio data processed by the DSP audio processing module.
The phase processing module comprises a relative phase measurer, a phase discriminator, a loop filter, a high-frequency VCO (voltage controlled oscillator), a synchronous frequency divider and an I2S clock and data phase rearrangement module;
data is input into an I2S clock and Data phase rearrangement module; the I2S clock and Data phase rearrangement module outputs three paths of rearranged Data of BCK, WCK and Data;
the BCK and the WCK of the audio frequency are input into a relative phase measurer and an I2S clock and data phase rearrangement module, meanwhile, one branch of the WCK is input into a phase discriminator, the phase discriminator outputs to a loop filter, the loop filter outputs to a high-frequency VCO, the high-frequency VCO outputs to a synchronous frequency divider and the relative phase measurer, and the synchronous frequency divider outputs to the phase discriminator; therefore, the phase discriminator, the loop filter, the high-frequency VCO and the synchronous frequency divider form a loop phase-locked frequency synthesizer to generate a high-frequency multiplication clock signal for phase quantization measurement, and the clock is synchronous with the WCK;
the relative phase measurer tests the relative phase of the WCK and the BCK by taking the rising edge of the WCK as a standard, and sends a quantized phase error value to the I2S clock and data phase rearrangement module to realize the relative rearrangement of the clock and data phases.
Taking the rising edge of the BCK clock signal as the establishment reference of the latch signal, and aligning the BCK clock signal to the centers of the rising edge and the falling edge of the Data signal when arranging the phase; the effective edge of the latch clock is established after the data is stable, and the data is not influenced by jitter and circuit delay dispersion.
The high frequency VCO is set to 16 times the BCK frequency.
The invention has the beneficial effects that:
the invention is provided with an over-frequency controller and an anti-shake processing module, wherein the over-frequency controller acquires a delay change curve and performs time integration, and the time range of the integration is t from the current moment 1 Integrating to the current time; inputting the integration result D into an adjusting model, and calculating the overclocking frequency f corresponding to different D values in the adjusting model; the overclocking controller obtains an overclocking frequency f and sends the overclocking frequency f to the DSP audio processing module, and the DSP audio processing module works according to the overclocking frequency f, so that the frequency conversion work of the DSP audio processing module is realized, and the delay is reducedThe method is particularly suitable for application scenes such as live broadcast, musical instrument playing and the like which need real-time feedback and have extremely high requirements on time delay.
The anti-shake processing module is used for generating anti-shake reverse phase signals and outputting the anti-shake reverse phase signals to the output interface, and the anti-shake reverse phase signals are superposed with signals output by the digital-to-analog conversion amplification module, so that an anti-shake effect is achieved. When the time delay is unstable, the volume of sound played by the loudspeaker can generate unstable jitter due to the unstable time delay, and in order to inhibit the jitter, an anti-jitter processing module is used to generate an inverted signal or a copied signal when the time delay is unstable, and the inverted signal or the copied signal is superposed with the original audio playing, so that the overlarge volume is reduced, the undersize volume is increased, and the anti-jitter effect is achieved.
Drawings
The accompanying drawings, which are included to provide a further understanding of the disclosed subject matter, are incorporated in and constitute a part of this specification. The drawings illustrate the implementations of the disclosed subject matter and, together with the detailed description, serve to explain the principles of implementations of the disclosed subject matter. No attempt is made to show structural details of the disclosed subject matter in more detail than is necessary for a fundamental understanding of the disclosed subject matter and various modes of practicing the same.
FIG. 1 is an overall architecture diagram of the present invention;
FIG. 2 is a schematic diagram of a phase processing module according to the present invention;
FIG. 3 is an input/output diagram of a phase processing module according to the present invention.
Detailed Description
The advantages, features and methods of accomplishing the same will be apparent from the drawings and the detailed description that follows.
Example 1:
referring to fig. 1, a digital audio anti-jitter and delay transmission prevention audio amplifier includes an input interface, an output interface, a DSP audio processing module, a digital-to-analog conversion and amplification module, a delay calculation module, a delay feedback module, an over-frequency controller, and an anti-jitter processing module;
the input interface is used for connecting a sound source, acquiring original audio data, and sending the original audio data to the DSP audio processing module;
the DSP audio processing module processes audio, and the processing comprises gain, distortion, EQ adjustment and reverberation increase;
the audio data output by the DSP is sent to a digital-to-analog conversion amplifying module, the digital-to-analog conversion amplifying module performs digital-to-analog conversion on the audio data to obtain an analog audio signal, and the analog audio signal is output to an output interface after being amplified;
the delay calculation module acquires the delay t between the input interface and the output interface and sends the delay to the delay feedback module; the delay feedback module sends the delay t to the overclocking controller; the over-frequency controller is connected with the DSP audio processing module and the anti-shake processing module;
the over-frequency control module is used for controlling the over-frequency performance of the DSP processing chip, the anti-shake processing module is used for generating anti-shake reversed-phase signals and outputting the anti-shake reversed-phase signals to the output interface, and the anti-shake reversed-phase signals are superposed with signals output by the digital-to-analog conversion amplification module, so that an anti-shake effect is achieved.
The input interface is provided with a marking module, the marking module is used for superposing a feature code on the original audio data acquired by the input interface, the feature code comprises a timestamp, and the feature code corresponds to a feature frequency after being subjected to digital-to-analog conversion; the feature code can be identified and analyzed by the delay calculation module, so that a time stamp of the feature code is obtained;
the feature codes are sent to the DSP audio processing module and then output to the digital-to-analog conversion and amplification module, and the feature codes are converted into analog signals and sent to an output interface;
and the delay calculation module acquires the characteristic frequency corresponding to the characteristic code from the output interface and calculates the time of the characteristic frequency, so that the delay delta t of the characteristic code which is output to the digital-to-analog conversion amplification module and transmitted to the output interface after passing through the DSP audio processing module is calculated.
The delay feedback module acquires the delay delta t obtained by the delay calculation module in real time; generating a time-delay delta t change curve along with time, namely a time-delay change curve, and sending the time-delay change curve to the overclocking controller;
the overclocking controller obtains a delay change curve and performs time integration, wherein the time range of the integration is t from the current moment 1 Is integrated toThe previous time; inputting the integration result D into an adjusting model, and calculating the overclocking frequency f corresponding to different D values in the adjusting model; the overclocking controller obtains overclocking frequency f and sends to DSP audio frequency processing module, and DSP audio frequency processing module is according to overclocking frequency f work to realize DSP audio frequency processing module frequency conversion work, reduce delay transmission.
The modeling method of the adjusting model comprises the following steps:
inputting the same section of audio signal into a DSP audio processing module, and performing single processing and superposition mixing processing by using four processing modes of gain, distortion, EQ adjustment and reverberation increase in the DSP audio processing module under basic working frequency, wherein the four processing modes are 15 combinations; a time delay calculating module is used for obtaining time delay delta t under 15 groups of different processing conditions 0 (ii) a Adjusting the working frequency of the DSP audio processing module to enable the DSP audio processing module to work in different over-frequency f states, and then obtaining delay delta t under 15 groups of different processing conditions under different frequencies by using a delay calculation module to obtain 15 groups of f-delta t curves;
let the delay time of each combination when the DSP audio processing module is working at the highest over-frequency be Deltat 1 (ii) a Calculating coefficient M = ([ Delta ] t) 0 -△t 1 )/△t 1 (ii) a The M value represents the sensitivity of the delay of the DSP audio processing module to the working frequency in different working modes, the larger the M value is, the more sensitive the M value is, and the more obvious the delay reduction realized by increasing the frequency is;
setting a threshold D in the tuning model 0 When D > D 0 Increasing the frequency Δ f = (k/M) · f 0 (ii) a When D is less than D 0 When the frequency is reduced, Δ f = (k/M) · f 0 (ii) a Wherein f is 0 To adjust the step size.
The anti-shake processing module acquires analog audio data from the digital-to-analog conversion and amplification module and uses the audio loudness A 0 The analog audio data is divided for the threshold value, when the actual loudness A of the analog audio data is larger than A 0 At the same time, the loudness exceeds A 0 When the actual loudness A of the analog audio data is less than A 0 Then, the corresponding part is copied, and the loudness after copying is adjusted to A 0 -A；
The delay feedback module acquires the delay delta t obtained by the delay calculation module in real time; generating a time-delay delta t change curve, namely a time-delay change curve, and sending the time-delay change curve to the anti-shake processing module;
the anti-shake processing module calculates t from the current moment to the front 1 Outputting the inverted audio and the copied audio generated by the anti-shake processing module to an output interface for superposition when the variance of delta t in the time period of the current moment exceeds a threshold value; and when the variance is smaller than the threshold value, outputting the reverse audio generated by the anti-shake processing module and the copied audio to an output interface for superposition.
Example 2:
the DSP audio processing module is also provided with a phase processing module; the phase processing module is connected between the DSP audio processing module and the digital-to-analog conversion amplification module and is used for performing phase processing on the audio data processed by the DSP audio processing module.
With reference to fig. 2-3, the phase processing module includes a relative phase measurer 5, a phase discriminator 1, a loop filter 2, a high-frequency VCO3, a synchronous frequency divider 4, and an I2S clock and data phase rearrangement module 6;
data is input into an I2S clock and Data phase rearrangement module 6; the I2S clock and Data phase rearrangement module 6 outputs three paths of rearranged Data of BCK, WCK and Data;
the BCK and the WCK of the audio source are input into a relative phase measurer 5 and an I2S clock and data phase rearrangement module 6, meanwhile, one branch of the WCK is input into a phase discriminator 1, the phase discriminator 1 outputs to a loop filter 2, the loop filter 2 outputs to a high-frequency VCO3, the high-frequency VCO3 outputs to a synchronous frequency divider 4 and the relative phase measurer 5, and the synchronous frequency divider 4 outputs to the phase discriminator 1; therefore, the phase discriminator 1, the loop filter 2, the high-frequency VCO3 and the synchronous frequency divider 4 form a loop phase-locked frequency synthesizer to generate a high frequency multiplication clock signal for phase quantization measurement, and the clock is synchronous with the WCK;
the relative phase measurer 5 tests the relative phase of the WCK and the BCK by taking the rising edge of the WCK as a standard, and sends the quantized phase error value to the I2S clock and data phase rearrangement module 6 to realize the relative rearrangement of the clock and data phases.
Taking the rising edge of the BCK clock signal as the establishment reference of the latch signal, and aligning the BCK clock signal to the centers of the rising edge and the falling edge of the Data signal when arranging the phase; the effective edge of the latch clock is established after the data is stable, and the data is not influenced by jitter and circuit delay dispersion.
The high frequency VCO3 is set to 16 times the BCK frequency.
The above description is only a preferred embodiment of the present invention, but the scope of the present invention is not limited thereto, and any person skilled in the art can easily think of the changes or substitutions within the technical scope of the present invention, and shall cover the scope of the present invention. Therefore, the protection scope of the present invention shall be subject to the protection scope of the appended claims.
Claims (5)
1. A digital audio anti-shake and delay transmission prevention audio amplifier is characterized by comprising an input interface, an output interface, a DSP audio processing module, a digital-to-analog conversion amplification module, a delay calculation module, a delay feedback module, an over-frequency controller and an anti-shake processing module;
the input interface is used for connecting a sound source, acquiring original audio data, and sending the original audio data to the DSP audio processing module;
the DSP audio processing module processes audio, and the processing comprises gain, distortion, EQ adjustment and reverberation increase;
the audio data output by the DSP is sent to a digital-to-analog conversion amplifying module, the digital-to-analog conversion amplifying module performs digital-to-analog conversion on the audio data to obtain an analog audio signal, and the analog audio signal is output to an output interface after being amplified;
the delay calculation module acquires the delay t between the input interface and the output interface and sends the delay to the delay feedback module; the delay feedback module sends the delay t to the over-frequency controller; the over-frequency controller is connected with the DSP audio processing module and the anti-shake processing module;
the over-frequency control module is used for controlling the over-frequency performance of the DSP processing chip, the anti-shake processing module is used for generating anti-shake reverse phase signals and outputting the anti-shake reverse phase signals to the output interface, and the anti-shake reverse phase signals are superposed with signals output by the digital-analog conversion amplification module, so that an anti-shake effect is achieved.
2. The digital audio anti-jitter and delay-tolerant audio amplifier of claim 1, wherein:
the input interface is provided with a marking module, the marking module is used for superposing a feature code on the original audio data acquired by the input interface, the feature code comprises a timestamp, and the feature code corresponds to a feature frequency after being subjected to digital-to-analog conversion; the feature code can be identified and analyzed by the delay calculation module, so that a time stamp of the feature code is obtained;
the feature codes are sent to the DSP audio processing module and then output to the digital-to-analog conversion amplification module, and are converted into analog signals to be sent to an output interface;
and the delay calculation module acquires the characteristic frequency corresponding to the characteristic code from the output interface and calculates the time of the characteristic frequency, so that the delay delta t of the characteristic code which is output to the digital-to-analog conversion amplification module and transmitted to the output interface after passing through the DSP audio processing module is calculated.
3. The digital audio anti-jitter and delay-tolerant audio amplifier of claim 2, wherein:
the delay feedback module acquires the delay delta t obtained by the delay calculation module in real time; generating a time delay delta t change curve along with time, namely a time delay change curve, and sending the time delay change curve to the over-frequency controller;
the overclocking controller obtains a delay change curve and performs time integration, wherein the time range of the integration is t from the current moment 1 Integrating to the current time; inputting the integration result D into an adjusting model, and calculating the overclocking frequency f corresponding to different D values in the adjusting model; the overclocking controller obtains overclocking frequency f and sends to DSP audio frequency processing module, and DSP audio frequency processing module is according to overclocking frequency f work to realize DSP audio frequency processing module frequency conversion work, reduce delay transmission.
4. The digital audio anti-jitter and anti-delay transmission audio amplifier of claim 3, wherein:
the modeling method of the adjusting model comprises the following steps:
inputting the same section of audio signals into a DSP audio processing module, and performing single processing and superposition mixing processing by using four processing modes of gain, distortion, EQ adjustment and reverberation in the DSP audio processing module under basic working frequency, wherein the four processing modes are 15 combinations in total; the time delay delta t under 15 groups of different processing conditions is obtained by using a time delay calculation module 0 (ii) a Adjusting the working frequency of the DSP audio processing module to enable the DSP audio processing module to work in different over-frequency f states, and then obtaining 15 groups of delay delta t under different processing conditions by using a delay calculation module to obtain 15 groups of f-delta t curves;
let the delay of each combination when the DSP audio processing module is operated at the highest over-frequency be Deltat 1 (ii) a Calculating coefficient M = ([ Delta ] t) 0 -△t 1 )/△t 1 (ii) a The M value represents the sensitivity of the delay of the DSP audio processing module to the working frequency in different working modes, the larger the M value is, the more sensitive is, and the more obvious the delay reduction realized by increasing the frequency is;
setting a threshold D in the tuning model 0 When D > D 0 Then, the frequency Δ f = (k/M) · f is increased 0 (ii) a When D is less than D 0 When the frequency is reduced, Δ f = (k/M) · f 0 (ii) a Wherein f is 0 To adjust the step size.
5. The digital audio anti-jitter and anti-delay transmission audio amplifier of claim 2, wherein the digital audio anti-jitter and anti-delay transmission audio amplifier is further characterized by
The anti-shake processing module obtains analog audio data from the digital-to-analog conversion amplifying module and uses the audio loudness A 0 The analog audio data is divided for the threshold value, when the actual loudness A of the analog audio data is larger than A 0 At the same time, the loudness exceeds A 0 When the actual loudness A of the analog audio data is less than A 0 Then, the corresponding part is copied, and the loudness after copying is adjusted to A 0 -A；
The delay feedback module acquires the delay delta t obtained by the delay calculation module in real time; generating a time-delay delta t change curve, namely a time-delay change curve, and sending the time-delay change curve to the anti-shake processing module;
the anti-shake processing module calculates t from the current moment to the front 1 Outputting the reversed phase audio generated by the anti-shake processing module and the copied audio to an output interface for superposition when the variance of delta t in the time period of the current moment exceeds a threshold value; and when the variance is smaller than the threshold value, outputting the reversed phase audio generated by the anti-shake processing module and the copied audio to the output interface for superposition.
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