CN106875953B - Method and system for processing analog mixed sound audio - Google Patents
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Abstract
The invention provides a method for processing analog mixed sound audio, which comprises the following steps: low-pass filtering the input first channel initial audio signal and second channel initial audio signal respectively, and forming a first channel analog early-reflecting audio signal and a second channel analog early-reflecting audio signal after delaying for a preset time period; superposing the first channel analog early-stage reflection audio signal and the second channel analog early-stage reflection audio signal, and then forming a first channel analog early-stage reverberation audio signal and a second channel analog early-stage reverberation audio signal after the analog obstacles influence the analog early-stage reflection audio signal through a plurality of FIR filter groups connected in series; respectively passing the first and second channel analog early-stage reflected audio signals through a plurality of cascaded IIR all-pass filter banks to form first and second channel analog late-stage reverberation audio signals; the first and second channel initial audio signals are proportionally mixed with the first and second channel simulated early reverberation audio signals and the first and second channel simulated late reverberation audio signals respectively to form a final simulated reverberation audio signal.
Description
Technical Field
The invention relates to the technical field of digital signal processing, in particular to an analog audio mixing audio processing method and system.
Background
A phenomenon in which sound waves propagate in a specific space and are reflected back and forth by obstacles such as walls, ceilings, and the like, and even after a sound source is stopped, a sound signal is left in the space for a while and disappears is called reverberation. Natural physical reverberation depends on a large number of conditions such as the size of a space, the shape of the space, the shape and material of an obstacle reflecting surface, and the humidity of air. Therefore, to obtain a specific reverberation effect, a specially designed building is required, and the building is often a building with a large scale, such as a concert hall, an opera house and the like.
The sound field experience generated by reverberation can give people an effect of putting a certain environment on the scene, and users very pursue complete immersive experience no matter in various games or various Karaoke shows, so the reverberation effect conforming to the picture environment can greatly enhance the user experience.
However, the room conditions required for natural physical reverberation are not present in the case and equipment of ordinary commercial KTV boxes, home karaoke players, passenger cars, personal cell phones, outdoor pull rod stereos, and the like. Therefore, there is a need to simulate the reverberation phenomenon of sound using computing devices, and especially after the theoretical basis of the digital processing of sound is perfected, a dedicated reverberation algorithm is necessary.
Disclosure of Invention
In order to solve the above problems, the present invention provides a method for processing a simulated mixed audio, which can simulate the reverberation effect of various environments. The audio processing method comprises the following steps:
low-pass filtering the input first channel initial audio signal and second channel initial audio signal respectively, and forming a first channel simulation early-stage reflection audio signal and a second channel simulation early-stage reflection audio signal after delaying for a preset time period;
superposing the first-channel simulated early-stage reflected audio signal and the second-channel simulated early-stage reflected audio signal, and then forming a first-channel simulated early-stage reverberation audio signal and a second-channel simulated early-stage reverberation audio signal after the simulated early-stage reflected audio signal is influenced by a simulated obstacle through a plurality of series-connected FIR filter groups;
respectively enabling the first channel simulation early-stage reflection audio signal and the second channel simulation early-stage reflection audio signal to pass through a plurality of cascaded IIR all-pass filter banks to form a first channel simulation later-stage reverberation audio signal and a second channel simulation later-stage reverberation audio signal;
and correspondingly and respectively proportionally mixing the first channel initial audio signal and the second channel initial audio signal with the first channel simulation early reverberation audio signal, the second channel simulation early reverberation audio signal, the first channel simulation late reverberation audio signal and the second channel simulation late reverberation audio signal to form a final simulation reverberation audio signal.
In one embodiment, according to the analog mixed audio processing method of the present invention, preferably, the FIR filter bank includes a plurality of single-pole low-pass filters connected in series, and a transfer function of the single-pole low-pass filters is as follows:
wherein the value of the coefficient band is adjustable and its absolute value is less than 1.
In one embodiment, according to the analog mixed audio processing method of the present invention, it is preferable that the value of the coefficient band is determined by the following formula:
wherein
fc＝e[-0.595435*log(d)+10.5189]
fsD is the given propagation distance, which is the sampling rate of the system.
In one embodiment, according to the analog mixed-tone audio processing method of the present invention, it is preferable that the IIR all-pass filter bank includes at least four IIR all-pass filters.
In one embodiment, according to the analog mixed-sound audio processing method of the present invention, it is preferable that the transfer function of the all-pass filter is:
in one embodiment, according to the analog mixed-sound audio processing method of the present invention, it is preferable that the signal after the cascaded all-pass filter bank is further superimposed on the analog early reflected audio signal as a part of an input signal for generating the analog late reverberation audio signal.
In one embodiment, according to the analog mixed audio processing method of the present invention, it is preferable that in the all-pass filter bank, each audio signal filtered by a single filter is proportionally output again through a gain amplifier, wherein a gain factor is adjustable.
In one embodiment, according to the analog mixed audio processing method of the present invention, it is preferable that, in the FIR filter bank, each audio signal after passing through each delay is proportionally output again through a gain amplifier, wherein a gain factor and a delay parameter are adjustable.
According to another aspect of the present invention, there is also provided an analog mixed audio processing system. The system comprises:
an audio device for generating and providing an initial audio signal and dividing the initial audio signal into a first channel audio signal and a second channel audio signal according to left and right ears arriving at a listening object;
the analog reverberation audio processing device is used for carrying out audio mixing processing on the received initial first channel audio signal and second channel audio signal through an early reflection simulation path, a late scattering simulation path and a late reverberation simulation unit to form an analog reverberation audio signal; and
an audio output device to output the formed analog reverberant audio signal;
in one embodiment, according to the analog mixed-sound audio processing system of the present invention, it is preferable that the analog reverberation audio processing device further includes an environment parameter adjusting interface for adjusting various parameters in the analog reverberation audio processing device according to the selected environment to be simulated.
The method and the system for processing the analog mixed sound audio have the advantages that the size of the analog reverberation space can be adjusted, and the parameters of the simulated obstacles can be adjusted. In addition, the reverberation density and the reverberation time length and the reverberation timbre can be adjusted.
More importantly, the method of the invention can be realized by using the device with an additional operating system as an application program, and is also suitable for being realized by using a DSP in an embedded device so as to realize high real-time performance, such as various karaoke devices, sound cards and the like.
Additional features and advantages of the invention will be set forth in the description which follows, and in part will be obvious from the description, or may be learned by practice of the invention. The objectives and other advantages of the invention will be realized and attained by the structure particularly pointed out in the written description and claims hereof as well as the appended drawings.
Drawings
In order to more clearly illustrate the embodiments of the present invention or the technical solutions in the prior art, the following briefly introduces the drawings required in the description of the embodiments or the prior art:
fig. 1 is a block diagram of a structure of a reverberation algorithm implementation commonly used in the prior art;
FIG. 2 shows a graph of the impulse response after reverberation using the prior art;
fig. 3 shows a flow chart of a reverberation algorithm implementation designed according to the principles of the present invention;
FIG. 4 shows a block diagram of a single-pole low-pass filter according to an embodiment of the invention;
FIGS. 5 and 6 show frequency response plots of the low pass filter for different parameters, respectively;
FIG. 7 shows a block diagram of an adjustable early reflection timbre FIR filter structure according to one embodiment of the present invention;
FIG. 8 shows a block diagram of a simple all-pass filter;
FIG. 9 shows a graph of the unit impulse response of the all-pass filter shown in FIG. 8;
FIG. 10 shows a block diagram of a system for nesting other systems in an all-pass filter;
FIG. 11 shows a block diagram of a practical all-pass filter nested with other systems;
FIG. 12 shows a graph of the unit impulse response of the all-pass filter shown in FIG. 11;
FIG. 13 shows a schematic diagram of a cascade structure of an all-pass filter according to an embodiment of the invention;
FIG. 14 shows a graph of the unit impulse response of the cascaded all-pass filter shown in FIG. 13;
FIG. 15 shows a block diagram of an algorithm for reverberation simulating audio according to an embodiment of the invention;
FIG. 16 shows a graph of the unit impulse response of the reverberation simulation algorithm as shown in FIG. 15, an
Fig. 17 shows a schematic diagram of an interface provided by the present invention for adjusting the structural parameters of the algorithm to simulate different ambient reverberation effects.
Detailed Description
The following detailed description of the embodiments of the present invention will be provided with reference to the drawings and examples, so that how to apply the technical means to solve the technical problems and achieve the technical effects can be fully understood and implemented. It should be noted that, as long as there is no conflict, the embodiments and the features of the embodiments of the present invention may be combined with each other, and the technical solutions formed are within the scope of the present invention.
In the following description, for purposes of explanation, numerous specific details are set forth in order to provide a thorough understanding of the embodiments of the invention. It will be apparent, however, to one skilled in the art that the present invention may be practiced without some of these specific details or with other methods described herein.
Additionally, the steps illustrated in the flow charts of the figures may be performed in a computer system such as a set of computer-executable instructions and, although a logical order is illustrated in the flow charts, in some cases, the steps illustrated or described may be performed in an order different than here.
Fig. 1 shows a block diagram of a reverberation algorithm implementation commonly used in the prior art. As shown in fig. 1, the initial audio signals of the left and right channels are input to a first delay filter 101 and a second delay filter 102, respectively, for processing. And simultaneously feeding back the left channel audio signal which is output after delay to the input of the first delay filter and the input of the second delay filter. These feedback signals are typically scaled up or down according to the results of the debugging to determine their effect on the analog audio signal. In the same principle, the right channel audio signal that has been delayed in output is fed back to the input of the first delay filter and to the input of the second delay filter.
As shown in fig. 1, the left channel original audio signal may pass through a delay device of, for example, 176ms, and then be amplified in the same scale to be output as a channel of audio signal simulating reverberation. While the left channel original audio signal passes through a delay device having a longer delay time of, for example, 246ms, and is then directly output.
A gain factor of 0.5 may be selected when feeding itself back to influence the analog audio signal. The analog left and right channel audio signals are then passed through a low pass filter (e.g., 0-6kHz signals) and a high pass filter (e.g., 6kHz or higher signals), respectively. When passing the high pass filter, the gain factor may be selected to be less than 1, for example, 0.4, to reduce the effect of the high frequency signal on the mixing.
The advantage of this configuration shown in fig. 1 is that less computing power and memory space is required, facilitating implementation on a variety of computing devices. However, this structure still has a great disadvantage in the effect of reverberation simulation.
Fig. 2 shows an impulse response diagram of a reverberation algorithm of this structure. As can be seen from the figure, the reverberation effect is low in density, strong in granular sensation and loud in metal sound. Moreover, the algorithm does not have an interface for adjusting the tone, and cannot simulate the difference of the absorption capacity of various barrier materials in the space to sound waves.
Therefore, the invention provides a method for processing simulated mixed sound audio, which can simulate the sound effects of various environments. As shown in fig. 3, there is shown a flow chart of a reverberation algorithm implementation designed according to the principles of the present invention.
In fig. 3, the workflow of the entire algorithm is shown. The algorithm uses two-channel stereo audio as the excitation input. In step S301, the input first-channel initial audio signal and second-channel initial audio signal are low-pass filtered and delayed for a predetermined time period, so as to form a first-channel simulated early-reflected audio signal and a second-channel simulated early-reflected audio signal that simulate propagation through an air medium.
This step is mainly used to simulate early reflections in the airborne process. The early reflected signal paths are: first, the audio signal passes through a single-pole low-pass filter, which is a low-pass filter that simulates the intensity attenuation of the signal as it travels through air. And then input to the adder after passing through the pre-delay buffer space. This path represents the first reflection of the sound wave from the left channel, which is also the strongest primary reflected signal. The right channel is also processed in the same structure, but the path parameters of the right channel may not be consistent with those of the left channel, so as to distinguish different spatial positions and propagation paths of the left and right channels.
Next, the influence of the distribution of obstacles in space on the early reflections is simulated in step S302. Specifically, the first channel analog early-stage reflected audio signal and the second channel analog early-stage reflected audio signal are superposed and then pass through a plurality of FIR filter banks connected in series to form a first channel analog early-stage reverberation audio signal and a second channel analog early-stage reverberation audio signal after the analog obstacles affect the analog early-stage reflected audio signal.
The left and right channels are overlapped together after first spatial reflection and are propagated together, and FIR filters are connected in series on a propagation path. By adjusting the parameters of the filter, the distribution combination of obstacles in the space and the absorption capacity of different obstacles to sound wave signals can be simulated. By adjusting various parameters of the FIR filter bank, the timbre of the reverberation can also be adjusted.
Then, in step S303, a sustain reverberation of the reverberation is simulated. Specifically, a first channel simulation early reflection audio signal and a second channel simulation early reflection audio signal are respectively passed through a plurality of cascaded IIR all-pass filter banks to form a first channel simulation late reverberation audio signal and a second channel simulation late reverberation audio signal.
Since the sound is already divided into many parts after early reflection. The early reflected signals are processed by an IIR all-pass filter bank formed in a cascading mode, so that the sound is exponentially multiplied in a fission-like mode, and the density and attenuation uniformity of reverberation are greatly enhanced. The number of the cascade connection of the IIR all-pass filters can be properly increased and decreased according to different computing capabilities of selected computing platforms, and the principle is that more than 4 are kept. Of course, the more filters in series, the more reverberant the reverberation is.
Finally, in step S304, the influence of the obstacle distribution in the simulation space on reverberation. Specifically, a first channel initial audio signal and a second channel initial audio signal are proportionally mixed with a first channel simulation early reverberation audio signal, a second channel simulation early reverberation audio signal, a first channel simulation late reverberation audio signal and a second channel simulation late reverberation audio signal respectively, so as to form a final simulation reverberation audio signal.
Obstacles in space actually affect the propagation path for each time. How this affects early reflections has been described above. The influence on the reverberation is realized by a structure similar to an FIR filter, the output of each stage of IIR all-pass filter is summarized to the final output by scaling one coefficient, a plurality of IIR all-pass filters are cascaded, the output and the corresponding coefficient form an FIR filter with larger time delay span, and parameters such as the timbre of the reverberation, the closing degree of the space and the like can be changed by changing the system.
The implementation of each simulation phase is described in detail below. As shown in fig. 4, the attenuation of sound propagating in air is simulated by a single-pole filter structure. The transfer function of the single-pole low-pass filter is as follows:
wherein the value of the coefficient band is adjustable and its absolute value is less than 1. As can be seen from the above equation, the filter has a zero and a pole, and for the stability of the system, it is necessary to ensure that the absolute value of the band parameter is less than 1. Under the condition, different low-pass characteristics can be generated by adjusting the value of the band, so that the absorption of sound signals under different spatial conditions is simulated.
Fig. 5 and 6 show the frequency response curves when band is equal to 0.2 and 0.8, respectively.
After a large number of experimental summaries, an empirical formula between the propagation distance and the band coefficient is obtained, as shown in the following formula. Thus the band coefficient can be determined according to the size of the space to be simulated and the position of the generating source, fs is the sampling rate of the system, and generally 48kHz is used. Here, d denotes a given distance.
Wherein f isc＝e[-0.595435*log(d)+10.5189](3)
Next, the simulation of early reflections is started. Early reflections are reflections of a sound source by relatively large reflectors in space, and have both causal relationships (represented as cascade relationships in the structure) and parallel relationships (represented as FIR taps in the figure) during each reflection, as shown in fig. 7. Preferably, the FIR filter bank comprises a plurality of single-pole low-pass filters connected in series. The strength of the signal of each path is determined by the size of the TAP coefficient TAP of the FIR. Let s [ N ] be the signal of the Nth stage, and h (N) be the impulse response of its attenuation model, which can be obtained from the previous transfer function. The tap coefficients of n stages are denoted by An, the total output can be expressed as follows:
and then simulating late scattering. The later scattering affects the timbre of the tail tone, the denser the later scattering, the better the monotonicity of the amplitude envelope, the better the listening feeling, and an all-pass filter is used in the method of generating the later scattering. The structure of a simple all-pass filter is shown in fig. 8.
In FIG. 8, if W-nIs the delay of k samples, its Z transformation (transfer function) is as follows, its unit impulse response is as shown in fig. 9, | h (Z) modulo, etcAt 1. It follows that its frequency response is flat, so called an all-pass filter, but it spreads out many copies of the signal in the time domain.
However, the simple all-pass filter described above is not sufficient to produce a sufficiently true scatter signal. In practical use, an all-pass filter is used that is improved over a basic all-pass filter. As shown in fig. 10, the other systems g (z) are nested in the all-pass filter.
The overall transfer function after nesting becomes shown in the following formula. Due to the all-pass filter, the frequency response is determined by the internal g (z):
according to the analog mixed-tone audio processing method of the present invention, it is preferable that the IIR all-pass filter bank includes at least four IIR all-pass filters. As shown in fig. 11, after the nesting of the above structures, not only can more dense sound copies be created during scattering, but also the low-pass attenuation caused by air during late scattering can be simulated, the transfer function h (z) of the above structures is as follows,
its unit impulse response is shown in fig. 12.
The scattered signals also experience multiple reflections and the paths to the last signal are more and more dense, and the causal relationship between the scattered signals is parallel, and the signals are also subjected to timbre adjustment by an FIR filter. As shown in fig. 13.
Let s [ N ] be the signal of the Nth stage, and h (N) be the impulse response of its attenuation model, which can be obtained from the previous transfer function. The tap coefficients of n stages are denoted by tapn, the total output can be expressed as follows:
where the sign-represents convolution.
If the stimulus is a unit pulse signal, the output of S2_ Out, i.e., the impulse response of the cascade structure, is shown in FIG. 14.
Assuming that the original input signal is signalin and the final output signal is SignalOut, the final input and output can be expressed by the following equations:
SignalOut＝q0*Signal+q1*S1out+q2*S2out (9)
q0 represents the proportion of the original signal in the final output signal, in which case q0 is not 0 since some of the signal is conducted through the body, and in other cases the value is 0, as is the case when the recipient of the simulated sound is also the originator of the sound. q1 denotes the proportion of early reverberation and q2 denotes the proportion of late scattered signals.
As shown in fig. 15, a detailed algorithm block diagram for analog mixed audio processing according to the principles of the present invention is shown.
Fig. 16 shows an impulse response diagram of the algorithm structure of fig. 15, which is a unit impulse response of the reverberation algorithm using six IIR all-pass filters. Compared with the prior art, the simulation effect graph has the advantages that the reverberation effect density is high, the granular sensation is weak, and the metal sound is weakened, so that the simulated reverberation effect is more vivid and comfortable.
In addition, as can be seen from fig. 15, according to the analog mixed-sound audio processing method of the present invention, the signal after the cascaded all-pass filter bank is further superimposed on the analog early reflected audio signal as a part of the input signal for generating the analog late reverberation audio signal.
Furthermore, in the all-pass filter bank, each audio signal filtered by a single filter is again outputted in proportion by a gain amplifier, wherein the gain factor is an adjustable parameter.
In one embodiment, according to the analog mixed audio processing method of the present invention, it is preferable that, in the FIR filter bank, each audio signal after passing through each delay is proportionally output again through a gain amplifier, wherein a gain factor and a delay parameter are adjustable.
The present invention also provides an interface for modifying these parameters to change the simulated environment, as shown in FIG. 17. In which simulation parameter selection in various environments such as cubicle, bathroom, large room, gym, auditorium, church, canyon, etc. is provided. Therefore, the size of the simulated reverberation space can be adjusted, and the simulated obstacle parameter can be adjusted according to the invention. In addition, the reverberation density and the reverberation time length and the reverberation timbre can be adjusted.
The method of the present invention is described as being implemented in a computer system. The computer system may be provided in a control core processor, for example. For example, the methods described herein may be implemented as software executable with control logic that is executed by a CPU in a control system. The functionality described herein may be implemented as a set of program instructions stored in a non-transitory tangible computer readable medium. When implemented in this manner, the computer program comprises a set of instructions which, when executed by a computer, cause the computer to perform a method capable of carrying out the functions described above. Programmable logic may be temporarily or permanently installed in a non-transitory tangible computer-readable medium, such as a read-only memory chip, computer memory, disk, or other storage medium. In addition to being implemented in software, the logic described herein may be embodied using discrete components, integrated circuits, programmable logic used in conjunction with a programmable logic device such as a Field Programmable Gate Array (FPGA) or microprocessor, or any other device including any combination thereof. All such embodiments are intended to fall within the scope of the present invention.
According to another aspect of the present invention, there is also provided an analog mixed audio processing system. The system comprises:
an audio device for generating and providing an initial audio signal and dividing the initial audio signal into a first channel audio signal and a second channel audio signal according to left and right ears arriving at a listening object;
the analog reverberation audio processing device is used for carrying out audio mixing processing on the received initial first channel audio signal and second channel audio signal through an early reflection simulation path, a late scattering simulation path and a late reverberation simulation unit to form an analog reverberation audio signal; and
an audio output device to output the formed analog reverberant audio signal;
in one embodiment, according to the analog mixed-sound audio processing system of the present invention, it is preferable that the analog reverberation audio processing device further includes an environment parameter adjusting interface for adjusting various parameters in the analog reverberation audio processing device according to the selected environment to be simulated.
The method and the system for processing the analog mixed sound audio have the advantages that the size of the analog reverberation space can be adjusted, and the parameters of the simulated obstacles can be adjusted. In addition, the reverberation density and the reverberation time length and the reverberation timbre can be adjusted.
More importantly, the method of the invention can be used in equipment with an additional operating system as an application program, and is also suitable for being realized in embedded equipment by using a DSP (digital signal processor) so as to realize high real-time performance, such as various karaoke devices, sound cards and the like.
It is to be understood that the disclosed embodiments of the invention are not limited to the particular structures or process steps disclosed herein, but extend to equivalents thereof as would be understood by those skilled in the relevant art. It is also to be understood that the terminology used herein is for the purpose of describing particular embodiments only, and is not intended to be limiting.
Reference in the specification to "one embodiment" or "an embodiment" means that a particular feature, structure, or characteristic described in connection with the embodiment is included in at least one embodiment of the invention. Thus, the appearances of the phrase "one embodiment" or "an embodiment" in various places throughout this specification are not necessarily all referring to the same embodiment.
While the above examples are illustrative of the principles of the present invention in one or more applications, it will be apparent to those of ordinary skill in the art that various changes in form, usage and details of implementation can be made without departing from the principles and concepts of the invention. Accordingly, the invention is defined by the appended claims.
Claims (10)
1. An analog mixed audio processing method, comprising:
low-pass filtering the input first channel initial audio signal and second channel initial audio signal respectively, and forming a first channel simulation early-stage reflection audio signal and a second channel simulation early-stage reflection audio signal after delaying for a preset time period;
superposing the first-channel simulated early-stage reflected audio signal and the second-channel simulated early-stage reflected audio signal, and then passing through a plurality of FIR filter groups connected in series to form an influence result of a simulated obstacle on superposition propagation of the first-channel simulated early-stage reflected audio signal and the second-channel simulated early-stage reflected audio signal, so that the current influence result is respectively used as a first-channel simulated early-stage reverberation audio signal and a second-channel simulated early-stage reverberation audio signal;
respectively enabling the first channel simulation early-stage reflection audio signal and the second channel simulation early-stage reflection audio signal to pass through a plurality of cascaded IIR all-pass filter banks to form a first channel simulation later-stage reverberation audio signal and a second channel simulation later-stage reverberation audio signal;
and correspondingly and respectively proportionally mixing the first channel initial audio signal and the second channel initial audio signal with the first channel simulation early reverberation audio signal, the second channel simulation early reverberation audio signal, the first channel simulation late reverberation audio signal and the second channel simulation late reverberation audio signal to form a final simulation reverberation audio signal.
2. An analog mixed audio processing method according to claim 1, wherein the FIR filter bank includes a plurality of single-pole low-pass filters connected in series, and a transfer function of the single-pole low-pass filters is as follows:
wherein the value of the coefficient band is adjustable and its absolute value is less than 1.
3. An analog mixed audio processing method according to claim 2, wherein the value of the coefficient band is determined by the following formula:
wherein
fc＝e[-0.595435*log(d)+10.5189]
fsD is the given propagation distance, which is the sampling rate of the system.
4. An analog mixed audio processing method according to claim 1, characterized in that the IIR all-pass filter bank includes at least four IIR all-pass filters.
5. An analog mixed audio processing method according to claim 4, wherein the transfer function of the all-pass filter is:
6. an analog mixed audio processing method according to claim 1, wherein the signal after the cascade of all-pass filter banks is further superimposed on the analog early reflected audio signal as a part of an input signal for generating the analog late reverberation audio signal.
7. An analog mixed audio processing method according to claim 1, wherein in the all-pass filter bank, each audio signal filtered by a single filter is again outputted in proportion by a gain amplifier, wherein a gain factor is adjustable.
8. An analog mixed audio processing method according to claim 1, wherein in the FIR filter bank, each audio signal after passing through each delay is again outputted in proportion by a gain amplifier, wherein a gain factor and a delay parameter are adjustable.
9. An analog mixed audio processing system, the system comprising:
an audio device for generating and providing an initial audio signal and dividing the initial audio signal into a first channel audio signal and a second channel audio signal according to left and right ears arriving at a listening object;
analog reverberation audio processing means to perform audio mixing processing of the received initial first and second channel audio signals through the early reflection simulation path, the late scattering simulation path and the late reverberation simulation unit by the method of any of claims 1-8 to form analog reverberation audio signals; and
an audio output device to output the formed analog reverberant audio signal.
10. The analog mixed sound audio processing system of claim 9, wherein the analog reverberation audio processing device further comprises an environment parameter adjusting interface for adjusting various parameters of the analog reverberation audio processing device according to the selected environment to be simulated.
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| CN110189740A (en) * | 2019-06-24 | 2019-08-30 | 上海矽诺微电子有限公司 | Reverberation circuit and its control method, chip and device |
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| CN112652290B (en) * | 2020-12-14 | 2023-01-20 | 北京达佳互联信息技术有限公司 | Method for generating reverberation audio signal and training method of audio processing model |
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