WO2017121245A1 - Procédé permettant de réaliser un son ambiophonique, dispositif électronique et support d'informations - Google Patents

Procédé permettant de réaliser un son ambiophonique, dispositif électronique et support d'informations Download PDF

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WO2017121245A1
WO2017121245A1 PCT/CN2016/113113 CN2016113113W WO2017121245A1 WO 2017121245 A1 WO2017121245 A1 WO 2017121245A1 CN 2016113113 W CN2016113113 W CN 2016113113W WO 2017121245 A1 WO2017121245 A1 WO 2017121245A1
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audio data
channel audio
channel
value
amplitude value
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Chinese (zh)
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杨将
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Tencent Technology Shenzhen Co Ltd
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Tencent Technology Shenzhen Co Ltd
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S1/00—Two-channel systems

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  • the present invention relates to the field of audio technologies, and in particular, to a surround sound implementation method, an electronic device, and a storage medium.
  • Surround sound is a special sound effect.
  • the sound field generated by surround sound preserves the sound source direction of the original signal, which gives the listener a strong sense of space and can realistically reproduce the spatial reverberation process of the performance hall. It has a more touching sense of presence.
  • surround sound is usually required to use the simulated head recording method during recording.
  • the simulated head recording method is to place two miniature omnidirectional microphones in the ear canal of a simulated human head which is basically consistent with the human head (close to the eardrum of the human ear). ), simulating the entire process of recording the human ear.
  • the current recording process of the human head recording method is complicated and costly; and the artificial head recording method is a special processing taken during recording, and the ordinary audio data cannot be surrounded by this method because the artificial head recording method is not used.
  • Stereo poor universality.
  • a surround sound implementation method an electronic device, and a storage medium are provided.
  • a surround sound implementation method comprising:
  • the first channel audio data and the second channel audio data are output through different sounding units, respectively.
  • An electronic device comprising a memory and a processor, the memory storing computer readable instructions, wherein the computer readable instructions are executed by the processor such that the processor performs the following steps:
  • the first channel audio data and the second channel audio data are output through different sounding units, respectively.
  • One or more computer readable non-volatile storage media storing computer readable instructions, when executed by one or more processors, cause the one or more processors to perform the steps of:
  • the first channel audio data and the second channel audio data are output through different sounding units, respectively.
  • FIG. 1 is a structural diagram and an application environment diagram of an electronic device for implementing a surround sound implementation method in an embodiment
  • FIG. 2 is a schematic flow chart of a method for implementing surround sound in an embodiment
  • FIG. 3 is a schematic diagram showing changes in the position of a sound image in a human brain when the time difference between the two ears changes in one embodiment
  • FIG. 4 is a schematic diagram of performing sound image splitting processing on first channel audio data and second channel audio data in one embodiment
  • FIG. 5 is a schematic diagram showing an equivalent circuit for performing weighting enhancement processing on the first channel audio data and the second channel audio data after the sound image splitting process in one embodiment
  • FIG. 6 is a flow chart showing the steps of compressing the amplitude values of the first channel audio data and the second channel audio data to a range of effective amplitude values in one embodiment
  • FIG. 7 is a structural block diagram of an electronic device in an embodiment
  • FIG. 8 is a structural block diagram of an amplitude value adjustment module in an embodiment
  • FIG. 9 is a structural block diagram of an electronic device in another embodiment.
  • FIG. 10 is a structural block diagram of an electronic device in still another embodiment.
  • an electronic device includes a processor coupled through a system bus, a non-volatile storage medium, an internal memory, and an audio output interface.
  • the processor has a computing function and a function of controlling the operation of the electronic device, the processor being configured to perform a surround sound implementation method.
  • Non-volatile storage media include magnetic storage media, optical storage media, and flash memory At least one of the storage mediums, the non-volatile storage medium storing an operating system.
  • the audio output interface is used for outputting an analog signal of audio data, and the sound signal can be converted into sound waves through a sounding unit connected to the audio output interface, so that the human ear can hear the sound content recorded by the audio data.
  • the electronic device can be a mobile terminal such as a mobile phone, a tablet computer, a music player, or a personal digital assistant (PDA), or can be a desktop computer.
  • PDA personal digital assistant
  • a surround sound implementation method is provided. This embodiment is exemplified by the method applied to the electronic device in FIG. 1 described above. As shown in FIG. 2, the method specifically includes the following steps:
  • Step 202 Acquire first channel audio data.
  • the electronic device acquires the first channel audio data from the audio data source
  • the audio data source may be stored locally in the electronic device, that is, the terminal may obtain the first channel audio data locally from the electronic device; the audio data source may also be stored in the network.
  • the electronic device can acquire the first channel audio data from the audio data source through the network.
  • the audio data source may use an audio format such as MP3 (Moving Picture Experts Group Audio Layer III), WMA (Windows Media Audio) or APE (a lossless audio format).
  • Step 204 Acquire second channel audio data having a fixed delay compared to the first channel audio data.
  • the first channel audio data and the second channel audio data are used to distinguish audio data of different channels. If the first channel audio data is left channel audio data, the second channel audio data may be right channel audio data; or, if the first channel audio data is right channel audio data, second channel audio data It can be left channel audio data.
  • the acquired second channel audio data is delayed by a fixed delay than the first channel audio data, and the fixed delay is an Interdural Time Difference (ITD), and the fixed delay is used to widen the sound field, specifically for splitting. Sound image to broaden the sound field.
  • the sound image is a human perception sound source, and the fixed delay is used to split the sound image, which means that if the first channel audio data and the second channel audio data with a fixed delay are respectively converted into sound waves, the sound is output to the human ear. People will perceive that there are two sound sources with the same or similar sound content but different positions in their heads.
  • the sound image position is moved from the center of the human brain along the axial direction between the ears.
  • the time difference between the two ears changes from 0.6ms to 10ms, the position of the sound image no longer moves along the axial direction between the ears, but the shape changes, resulting in widening of the sound image, and the variation range increases with the increase of the time difference between the two ears.
  • the time difference between the two ears continues to increase to a certain value, the widened sound image produced in the human brain is split into two symmetrical and unwidened sound images.
  • the specific value here is generally between 15ms and 50ms, and the specific value is also related to the characteristics of the audio data source, such as the channel difference existing in the audio data source itself.
  • the fixed delay can take values between 15ms and 50ms.
  • Step 206 Adjust amplitude values of the first channel audio data and/or the second channel audio data such that the amplitude value of the first channel audio data is smaller than the amplitude value of the second channel audio data.
  • the electronic device may adjust the amplitude value of the first channel audio data, or adjust the amplitude value of the second channel audio data, or simultaneously adjust the amplitude value of the first channel audio data and the amplitude of the second channel audio data.
  • the amplitude value here refers to the time domain amplitude value. After adjustment, the amplitude value of the first channel audio data is smaller than the amplitude value of the corresponding second channel audio data.
  • Step 208 outputting first channel audio data and second channel audio data through different sounding units, respectively.
  • the electronic device can connect two sounding units, that is, a first sounding unit and a second sounding unit, and the two sounding units can be a left ear sounding unit and a right ear sounding unit of the earphone, respectively.
  • the electronic device can convert the first channel audio data into an analog signal, output the sound wave through the first sounding unit, convert the second channel audio data into an analog signal, and then generate the sound wave through the second sounding unit.
  • the second channel audio data has a fixed delay compared to the first channel audio data, and the fixed delay can split the sound image, so that the sound source perceived by the human ear is more than one, and the sound field is broadened. .
  • the fixed delay generates a bias phenomenon, and the amplitude value of the first channel audio data is smaller than the second channel audio data by adjusting the amplitude values of the first channel audio data and/or the second channel audio data. Amplitude value.
  • the amplitude value of the delayed second channel audio data is enhanced compared to the amplitude of the undelayed first channel audio data, and the bias caused by the delay is compensated by the change of the amplitude value, and the surround sound is realized. At the same time avoid the phenomenon of partial sound. It is not necessary to simulate the human head recording mode, and ordinary audio data can be realized by computer program processing, and has a strong general Fitness.
  • step 204 specifically includes: acquiring second channel audio data that is time synchronized with the first channel audio data, and inserting one frame of audio data into the time synchronized second channel audio data.
  • the electronic device can directly acquire the second channel audio data that is time-synchronized with the first channel audio data from the audio data source. If the audio data source itself has no channel distinction, the two channels can be obtained from the audio data source. The same audio data is used as time-synchronized first channel audio data and second channel audio data, respectively.
  • the length of time of one frame of audio data is generally between 15ms and 50ms of a fixed delay.
  • the audio data of one frame in the MP3 audio format is 26ms, so that the sound can be quickly realized by inserting one frame of audio data in the second audio data.
  • the fixed delay is the length of time of one frame of audio data.
  • z denotes a z-transformation, which can transform a time domain signal (ie, a discrete time series) into an expression in a complex frequency domain.
  • T represents a fixed delay
  • the inserted audio data can be generated according to the previous frame audio data and the subsequent frame audio data at the insertion point, so that the previous frame audio data smoothly transitions to the inserted one frame of audio data, and the inserted one frame of audio data smoothly transitions to the rear.
  • the inserted audio data may be generated according to the previous frame audio data and the subsequent frame audio data at the insertion point, specifically according to the last sample point value of the previous frame audio data and the first sample point value of the subsequent frame audio data. generate. This prevents noise from being generated by inserting one frame of audio data.
  • step 204 specifically includes: acquiring the same time as the first channel audio data.
  • the second channel audio data of the step deletes one frame of audio data in the first channel audio data.
  • the electronic device can not only insert one frame of audio data into the second channel audio data, but also delete one frame of audio data from the first channel audio data, so that the second channel audio data is compared with the first Channel audio data has a fixed delay.
  • the fixed delay is the length of time of one frame of audio data.
  • the length of time for one frame of audio data is generally between 15ms and 50ms for a fixed delay.
  • the audio data of one frame in the MP3 audio format is 26ms, which can be quickly deleted by deleting one frame of audio data in the first channel audio data. Realizes sound image splitting, and does not have too much impact on the original audio data, efficient and accurate.
  • the audio data of the previous frame of the deleted one frame of audio data and the audio data of the latter frame are smoothly transitioned.
  • the first sample point value of the deleted audio data and the last sample point value are equal or the difference satisfies the minimum condition; or, the previous frame audio of the deleted one frame of the audio data may also be
  • the data and the subsequent frame of audio data are processed such that the previous frame of audio data and the subsequent frame of audio data are smoothly transitioned. This prevents noise from being generated by deleting one frame of audio data.
  • the step 206 specifically includes: adding the first channel audio data to the first channel center gain value, where the first channel center gain value is the sum of the first channel audio data and the second channel audio data. Multiplying the first center coefficient; adding the second channel center data gain value to the second channel audio data, and the second channel center gain value is the sum of the first channel audio data and the second channel audio data Then multiplied by the second center coefficient; the first center coefficient is smaller than the second center coefficient.
  • the first centering coefficient may take one, and the corresponding second centering factor may take (1, 1.2). In one embodiment, the first centering factor is taken as 1 and the second centering factor is taken as 1.2.
  • Li is the first channel audio after the sound image splitting process.
  • Ri is the second channel audio data after the sound image split processing
  • Lo is the first channel audio data after the weight enhancement processing
  • Ro is the second channel audio data after the weight enhancement processing.
  • "-" means that the input signal is made worse
  • "+” means that the input signals are summed
  • the inverter is used. Reverse the phase of the signal that will pass.
  • n represents the center coefficient
  • p represents the spatial sense gain parameter
  • Chinese represents the head correlation transform function, and is a sound effect localization algorithm.
  • the weighting enhancement processing of the first channel audio data Li and the second channel audio data Ri after the sound image splitting processing may adopt the following formula (2):
  • Lo represents the first channel audio data that is output after the first channel audio data Li is subjected to the weight enhancement processing
  • Ro represents the second channel audio data that is output after the second channel audio data Ri is subjected to the weight enhancement processing.
  • n L represents the first mid-coefficient and n R represents the second mid-coefficient. Indicates that the convolution is sought.
  • the first centering coefficient n L may take 1 and the corresponding second center coefficient n R may take (1, 1.2). In one embodiment, the first center coefficient n L takes 1 and the second The center coefficient n R can be taken as 1.2.
  • the sound image is split into two: the original sound image and the rear sound image, while the sound field is pulled to both ears, leaving less sound field in the head.
  • the sound image field at both ears will spread to the surroundings, thus increasing the sound field distribution in the head; at the same time, giving the non-delay channel a small n value in the calculation can weaken the partial sound effect.
  • step 206 further comprising: performing high pass filtering and low pass filtering on the first channel audio data and the second channel audio data.
  • the electronic device may be filtered in the order of low-pass filtering and high-pass filtering first, or may be filtered in the order of high-pass filtering and low-pass filtering. Both high-pass filtering and low-pass filtering can be implemented by a computer program calling the corresponding function.
  • the extremely low frequency portion and the extremely high frequency portion of the audio do not affect the sound perception.
  • the audio data of the two channels are subjected to high and low cut processing to reduce high frequency sound and low frequency. The effect of the sound on the finite amplitude value and does not affect the original audio quality.
  • the electronic device may first filter the first channel audio data and the second channel audio data by using a low pass filter function, and then filter the first channel audio data and the second channel audio data by using a high pass filter function. It can be expressed by the following formula (3):
  • Li represents first channel audio data before high pass filtering and low pass filtering
  • Ri represents second channel audio data before high pass filtering and low pass filtering
  • LP() represents a low-pass filter function
  • HP() represents a high-pass filter function
  • Lo represents the first channel audio data after high-pass filtering and low-pass filtering
  • Ro represents the second channel audio data after high-pass filtering and low-pass filtering.
  • the method further includes the step of compressing the amplitude values of the first channel audio data and the second channel audio data into a range of effective amplitude values.
  • the method further includes the following steps:
  • Step 602 When the corresponding amplitude value of the first channel audio data or the second channel audio data exceeds the effective amplitude value range, the corresponding amplitude value is obtained according to the maximum effective amplitude value of the effective amplitude value range, and the segmentation value sequence is obtained.
  • the effective amplitude value range can be expressed as [-A, A], where A represents the maximum effective amplitude value and A can take 1.
  • Segmentation according to the maximum effective amplitude value means that the segmentation is performed in units of the maximum effective amplitude value, and the obtained segmentation values constitute a sequence of segmentation values in the order of segmentation. For example, assuming that the maximum effective amplitude value is 1, and the absolute value of the corresponding amplitude value is 3.2, the sequence of segmentation values obtained by segmentation according to the maximum effective amplitude value is 1, 1, 1, 0.2.
  • Step 604 Obtain a weight of each segment value in the sequence of segment values, and the obtained values of the weights are sequentially decremented and the sum of the obtained weights is less than or equal to 1.
  • the electronic device acquires a weight assigned to each segment value in the sequence of segment values, and the acquired weights are sequentially decremented in the order of the sequence of segment values, and the sum of the weights of all segment values in the sequence of segment values is smaller than Equal to 1.
  • the sum of the weights is less than or equal to 1 is that the weight is required to satisfy the condition, and does not mean that the sum of the weights is to be calculated.
  • the sequence of segmentation values is 1, 1, 1, 0.2; then the weights can be 0.5, 0.25, 0.1, 0.08 in turn, and these weights are decremented, and the sum is 0.93, satisfying the weight and the condition less than 1. .
  • Step 606 Calculate a weighted sum of the sequence of segment values according to the obtained weights.
  • Step 608 resetting the corresponding amplitude value according to the weighted sum.
  • the amplitude value of the reset should be the same as the sign of the corresponding amplitude value. If the corresponding amplitude value is originally a positive value, the corresponding amplitude value is reset to a weighted sum; if the corresponding amplitude value is originally a negative value, the corresponding amplitude value is reset to the opposite of the weighted sum.
  • the portion where the corresponding amplitude value does not exceed the effective amplitude value range and the portion exceeding the effective amplitude value range are respectively compressed by different compression ratios, so that the corresponding amplitude value after compression belongs within the effective amplitude value range.
  • the compression ratio of the portion exceeding the effective amplitude value range is larger than the compression ratio of the portion not exceeding the effective amplitude value range.
  • step 604 specifically includes: obtaining a weight parameter K greater than one; and sequentially selecting each of the segments in the sequence of segment values according to a ratio of 1-1/K and a ratio of 1/K. Segment values are assigned weights.
  • the weight parameter K is a control parameter used to control the weight allocation strategy. By adjusting the size of K, the weighting ratio of each segment value in the segment value sequence can be controlled. K is greater than 1, so 1-1/K is greater than 0, and 1/K is greater than 0, and all of the constituent geometric series are positive.
  • the weights are assigned to the segment values in the sequence of segment values in the order of 1-1/K and the ratio of 1/K, which may be any from the series of equals. From the position (for example, from the first item), the values in the equal series are taken continuously or intermittently to assign weights to the respective segment values in the sequence of segment values, and the assigned weights must satisfy the successive decrement and the sum of the weights is less than or equal to 1. condition.
  • an electronic device 700 includes a first obtaining module 701 , a second acquiring module 702 , an amplitude value adjusting module 703 , and an output module 704 .
  • the first obtaining module 701 is configured to acquire first channel audio data.
  • the first obtaining module 701 can be configured to locally acquire the first channel audio data from the electronic device, and can also be used to obtain the first channel audio data from the audio data source on the network.
  • the second obtaining module 702 is configured to acquire second channel audio data having a fixed delay compared to the first channel audio data.
  • the acquired second channel audio data is delayed by a fixed delay than the first channel audio data.
  • the fixed delay is ITD, and the fixed delay is used to widen the sound field, specifically for widening the sound field by splitting the sound image.
  • the sound image is a human perception sound source, and the fixed delay is used to split the sound image, which means that if the first channel audio data and the second channel audio data with a fixed delay are respectively converted into sound waves, the sound is output to the human ear. People will perceive that there are two sound sources with the same or similar sound content but different positions in their heads.
  • the amplitude value adjustment module 703 is configured to adjust amplitude values of the first channel audio data and/or the second channel audio data such that the amplitude value of the first channel audio data is smaller than the amplitude value of the second channel audio data.
  • the amplitude value adjustment module 703 can be used to adjust the amplitude value of the first channel audio data, or adjust the amplitude value of the second channel audio data, or simultaneously adjust the amplitude value of the first channel audio data and the second channel.
  • the amplitude value of the audio data here refers to the time domain amplitude value. After adjustment, the amplitude value of the first channel audio data is smaller than the amplitude value of the corresponding second channel audio data.
  • An output module 704 configured to output first channel audio data and respectively through different sounding units Second channel audio data.
  • the output module 704 may never convert the first channel audio data into an analog signal, and then output the sound wave through the first sounding unit, convert the second channel audio data into an analog signal, and then generate the sound wave through the second sounding unit.
  • the second channel audio data has a fixed delay compared to the first channel audio data, and the fixed time delay can split the sound image, so that the sound source perceived by the human ear is more than one, and the sound field is widened.
  • the fixed delay generates a bias phenomenon, and the amplitude value of the first channel audio data is smaller than the second channel audio data by adjusting the amplitude values of the first channel audio data and/or the second channel audio data. Amplitude value.
  • the amplitude value of the delayed second channel audio data is enhanced compared to the amplitude of the undelayed first channel audio data, and the bias caused by the delay is compensated by the change of the amplitude value, and the surround sound is realized. At the same time avoid the phenomenon of partial sound.
  • the second obtaining module 702 is further configured to acquire second channel audio data that is time-synchronized with the first channel audio data, and insert one frame of audio data into the time-synchronized second channel audio data.
  • the second acquisition module 702 can be configured to obtain second channel audio data that is time synchronized with the first channel audio data directly from the audio data source. If the audio data source itself does not have a distinction of channels, the first obtaining module 701 and the second obtaining module 702 may respectively obtain two identical audio data from the audio data source, respectively, as time-synchronized first channel audio data. And second channel audio data.
  • the length of time of one frame of audio data is generally between 15ms and 50ms of a fixed delay.
  • the audio data of one frame in the MP3 audio format is 26ms, so that the sound can be quickly realized by inserting one frame of audio data in the second audio data.
  • the fixed delay is the length of time of one frame of audio data.
  • the inserted audio data can be generated according to the previous frame audio data and the subsequent frame audio data at the insertion point, so that the previous frame audio data smoothly transitions to the inserted one frame of audio data, and the inserted one frame of audio data smoothly transitions to the rear.
  • the inserted audio data can be based on the front of the insertion point
  • One frame of audio data and the next frame of audio data are generated, which may be generated according to the last sample point value of the previous frame of audio data and the first sample point value of the subsequent frame of audio data. This prevents noise from being generated by inserting one frame of audio data.
  • the second obtaining module 702 is further configured to acquire second channel audio data that is time-synchronized with the first channel audio data, where the first acquiring module 701 is further configured to delete one in the first channel audio data.
  • Frame audio data is further configured to acquire second channel audio data that is time-synchronized with the first channel audio data.
  • the length of time for one frame of audio data is generally between 15ms and 50ms for a fixed delay.
  • the audio data of one frame in the MP3 audio format is 26ms, which can be quickly deleted by deleting one frame of audio data in the first channel audio data. Realizes sound image splitting, and does not have too much impact on the original audio data, efficient and accurate.
  • the audio data of the previous frame of the deleted one frame of audio data and the audio data of the latter frame are smoothly transitioned.
  • the first sample point value of the deleted audio data and the last sample point value are equal or the difference satisfies the minimum condition; or, the previous frame audio of the deleted one frame of the audio data may also be
  • the data and the subsequent frame of audio data are processed such that the previous frame of audio data and the subsequent frame of audio data are smoothly transitioned. This prevents noise from being generated by deleting one frame of audio data.
  • the amplitude value adjustment module 703 includes a first channel amplitude value adjustment module 703a and a second channel amplitude value adjustment module 703b.
  • the first channel amplitude value adjustment module 703a is configured to add the first channel center gain value to the first channel audio data, where the first channel center gain value is the first channel audio data and the second channel audio data. And multiply by the first mid-coefficient.
  • the second channel amplitude value adjustment module 703b is configured to add the second channel center gain value to the second channel audio data, and the second channel center gain value is the first channel audio data and the second channel audio.
  • the sum of the data is multiplied by the second center coefficient; the first center coefficient is smaller than the second center coefficient.
  • the sound image is split into two: the original sound image and the rear sound image, while the sound field is pulled to both ears, leaving less sound field in the head.
  • the sound image field at both ears will spread to the surroundings, thus increasing the sound field distribution in the head; The smaller n value of the non-delay channel can attenuate the partial sound effect.
  • the electronic device 700 further includes a high pass filtering module 705 and a low pass filtering module 706.
  • the high pass filtering module 705 is configured to perform high pass filtering on the first channel audio data and the second channel audio data
  • the low pass filtering module 706 is configured to perform low pass filtering on the first channel audio data and the second channel audio data.
  • the extremely low frequency portion and the extremely high frequency portion of the audio do not affect the sound perception.
  • the audio data of the two channels are subjected to high and low cut processing to reduce high frequency sound and low frequency. The effect of the sound on the finite amplitude value and does not affect the original audio quality.
  • the electronic device 700 further includes a segmentation module 707, a weight acquisition module 708, and an amplitude value assignment module 709.
  • the segmentation module 707 is configured to obtain, according to the maximum effective amplitude value of the effective amplitude value range, the corresponding amplitude value when the corresponding amplitude value of the first channel audio data or the second channel audio data exceeds the effective amplitude value range Sequence of values.
  • the weight obtaining module 708 is configured to obtain weights of the segment values in the sequence of segment values, and the obtained values of the weights are sequentially decremented and the sum of the obtained weights is less than or equal to 1.
  • the amplitude value assignment module 709 is configured to calculate a weighted sum of the sequence of segment values according to the obtained weights; and reset the corresponding amplitude values according to the weighted sum.
  • the weight obtaining module 708 and the amplitude value assigning module 709 may be included in an amplitude value pressure limiting module (not shown) for respectively respectively filtering a portion of the corresponding amplitude value that does not exceed the effective amplitude value range and a portion exceeding the effective amplitude value range. Compression of different compression ratios is performed such that the corresponding amplitude values after compression are within the range of effective amplitude values.
  • the compression ratio of the portion exceeding the effective amplitude value range is larger than the compression ratio of the portion not exceeding the effective amplitude value range
  • the weight obtaining module 708 is further configured to obtain a weight parameter K greater than 1; and the equal ratio sequence with the first item of 1-1/K and the ratio of 1/K is sequentially in the sequence of segment values. Each segment value is assigned a weight.
  • the weight parameter K is a control parameter used to control the weight allocation strategy. By adjusting the size of K, the weighting ratio of each segment value in the segment value sequence can be controlled. K is greater than 1, so 1-1/K is greater than 0, and 1/K is greater than 0, and all of the constituent geometric series are positive.
  • the segment values are assigned weights, which can ensure that the sum of the weights is less than 1, and can quickly assign appropriate weights to the segment values in the segmentation value sequence, which is very efficient.
  • the weight obtaining module 708 assigns weights to the segment values in the sequence of segment values in the order of 1-1/K and the ratio of 1/K. Starting from any position of the sequence, the values in the equal series are taken continuously or at intervals to assign weights to the respective segment values in the sequence of segment values, and the assigned weights necessarily satisfy the condition that the sum of the weights is successively decremented and the sum of the weights is less than or equal to one.
  • the storage medium may be a non-volatile storage medium such as a magnetic disk, an optical disk, a read-only memory (ROM), or a random access memory (RAM).

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Abstract

La présente invention concerne un procédé pour réaliser un son ambiophonique, un dispositif électronique et un support d'informations, ledit procédé consistant : à acquérir des données audio d'un premier canal ; à acquérir des données audio d'un second canal qui a un retard fixe par comparaison auxdites premières données audio de canal ; à régler la valeur d'amplitude desdites premières données audio de canal et/ou desdites secondes données audio de canal, de telle sorte que la valeur d'amplitude desdites premières données audio de canal est inférieure à la valeur d'amplitude desdites secondes données audio de canal ; et à exporter respectivement lesdites premières données audio de canal et lesdites secondes données audio de canal au moyen de différentes unités de production de son.
PCT/CN2016/113113 2016-01-14 2016-12-29 Procédé permettant de réaliser un son ambiophonique, dispositif électronique et support d'informations Ceased WO2017121245A1 (fr)

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CN201610025695.8A CN106973355B (zh) 2016-01-14 2016-01-14 环绕立体声实现方法和装置

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