KR102409697B1 - Audio synchronization system using unidirectional offset estimation - Google Patents

Abstract

The present invention relates to an audio synchronization method using one-way offset estimation for synchronizing an analog output and a digital output using an offset tracking value generated by a digital demodulator.

Description

Audio synchronization system using unidirectional offset estimation

The present invention relates to an audio synchronization system using one-way offset estimation, and more particularly, to an audio synchronization system using one-way offset estimation that synchronizes an analog output and a digital output using an offset tracking value generated from a digital demodulator.

1 is a diagram illustrating a conventional SL input/output audio synchronization system.

Assuming that audio input/output is 48ksps, audio offset occurs at XTAL (x1 offset) of RF and XTAL (x2 offset) of Audio DSP.

In the case of analog audio, since it is usually decoded in RF, it has 48ksps + x1 offset. In the case of digital audio, when the digital demodulation is performed from the input (point 2) of the tuner, the offset transmitted from the transmitter is generally completely removed. You can get 48ksps audio.

Thereafter, the time difference, signal level, and phase between the two audios are matched to be the same through SL (Seamless Linking), and one audio is delivered to the Audio DSP. At this time, this is compensated through SRC (Sample Rate Converter) in SL so that the sound is not interrupted by the x2 offset generated by Audio DSP.

However, in the conventional system, despite the SL linkage due to an offset difference between the two audios in the SL input, the two audios are distorted after a certain period of time, and in this case, audio drop in broadcasting causes a big problem.

For this, SRC control of SL output is possible only when the x2 offset generated by Audio DSP is known (system control is not known to Tier2, which supplies SL S/W to Tier 1 area, so either manual SRC processing or Tier1 directly SRC for SL output will be implemented), and in the case of Tier 1, which produces relatively low-cost products, only the continuity between Audio DSP and DAC is involved, and there is a limit in that it does not participate in the continuity between SL and Audio DSP.

On the other hand, the above-mentioned background art is technical information that the inventor possessed for the purpose of derivation of the present invention or acquired during the derivation process of the present invention, and it cannot be said that it is necessarily known technology disclosed to the general public before the filing of the present invention. .

Korean Patent Publication No. 10-2017-0103600

One aspect of the present invention provides an audio synchronization system using one-way offset estimation for synchronizing an analog output and a digital output using an offset tracking value generated from a digital demodulator.

The technical problems of the present invention are not limited to the technical problems mentioned above, and other technical problems not mentioned will be clearly understood by those skilled in the art from the following description.

An audio synchronization method using one-way offset estimation according to an embodiment of the present invention synchronizes an analog audio output and a digital audio output using an offset tracking value generated by a digital demodulator.

The audio synchronization method using the one-way offset estimation,

In order to remove the offset generated by the RF receiver, the first sampling rate converter provided at the input terminal of the SL (Seamless Linking) synchronizes the analog audio output and the digital audio output using the first offset tracking value generated by the digital demodulator. step;

An offset predicted to be generated by an audio DSP (Digital Signal Processor) is estimated based on the error between the input and output of the audio buffer, and a second offset tracking value for removing the estimated offset is provided at the output terminal of the SL. 2 sending to a sampling rate converter; and

and correcting the audio signal based on the second offset tracking value by a second sampling rate converter provided at the output terminal of the SL.

Estimating the offset generated by the audio DSP (Digital Signal Processor) based on the error between the input and output of the audio buffer,

generating underrun and overrun of the audio buffer by an error between the input and output of the audio buffer;

It is characterized in that the buffer emptying speed is converted into an error rate based on the underrun and overrun, and the second offset tracking value is generated based on the converted error rate.

According to one aspect of the present invention described above, unlike the conventional bidirectional offset removal method of applying the offset estimated from the audio DSP to the SRC, it is an audio synchronization method using the unidirectional offset estimation using the offset tracking value generated in the digital demodulator. Analog output and digital output can be synchronized.

In addition, by putting x2 offset tracking in the SL stage, it can be operated independently by removing the feedback circuit from the Audio DSP.

1 is a diagram illustrating a conventional audio input/output system.
2 is a diagram illustrating an audio synchronization system using unidirectional offset estimation according to an embodiment of the present invention.
3 to 5 are diagrams illustrating a detailed audio synchronization process using the audio synchronization system using one-way offset estimation according to the present invention.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS [0012] DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS [0010] DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS [0010] Reference is made to the accompanying drawings, which show by way of illustration specific embodiments in which the present invention may be practiced. These embodiments are described in sufficient detail to enable those skilled in the art to practice the present invention. It should be understood that the various embodiments of the present invention are different but need not be mutually exclusive. For example, certain shapes, structures, and characteristics described herein with respect to one embodiment may be implemented in other embodiments without departing from the spirit and scope of the invention. In addition, it should be understood that the location or arrangement of individual components within each disclosed embodiment may be changed without departing from the spirit and scope of the present invention. Accordingly, the detailed description set forth below is not intended to be taken in a limiting sense, and the scope of the present invention, if properly described, is limited only by the appended claims, along with all scope equivalents as those claimed. Like reference numerals in the drawings refer to the same or similar functions throughout the various aspects.

Hereinafter, preferred embodiments of the present invention will be described in more detail with reference to the drawings.

2 is a diagram showing a schematic configuration of an audio synchronization system using unidirectional offset estimation according to an embodiment of the present invention.

The audio synchronization system using one-way offset estimation according to the present invention is implemented in a radio receiver to synchronize the analog audio output and the digital audio output using the offset tracking value generated by the digital demodulator. A radio receiver is a device for receiving a heterogeneous audio signal including an analog signal such as FM and AM and a digital signal such as DAB (Digital Audio Broadcasting) and DRM (Digital Radio Mondiale).

Specifically, in the audio synchronization method using the unidirectional offset estimation according to an embodiment of the present invention, in order to remove the offset generated in the RF receiver, seamless linking (SL) using the first offset tracking value generated in the digital demodulator ) a first step of synchronizing the analog audio output and the digital audio output by a first sampling rate converter provided at the input terminal; An offset predicted to be generated by an audio DSP (Digital Signal Processor) is estimated based on the error between the input and output of the audio buffer, and a second offset tracking value for removing the estimated offset is provided at the output terminal of the SL. a second step of sending to a 2 sampling rate converter; and a third step of correcting the audio signal based on the second offset tracking value by a second sampling rate converter provided at the output terminal of the SL.

In the first step, the digital audio output is synchronized with the analog FM output or the FM output is synchronized with the digital audio output.

3 is a diagram illustrating an example of synchronizing a digital audio output to an analog FM output. Sample rate (SRC) of a digital stage using an XTAL ppm offset tracking value (first offset tracking value) generated from a digital demodulation. Converter, sampling rate converter) can be used to synchronize the two audios.

4 is a diagram showing an example of synchronizing the FM output with the digital audio output. Using the XTAL ppm offset tracking value (first offset tracking value) generated in digital demodulation, the two audios are synchronized through SRC processing of the analog stage. can

Here, ppm is an abbreviation of parts per million, and is a unit used to indicate parts per million, that is, what quantity occupies in parts per million. Therefore, in the crystal (Xtal), the error rate can be expressed as 0.01% = 100ppm, 0.005% = 50ppm, and 0.001% = 10ppm. In this case, when the ppm value is large, since one-time compensation causes a problem in audio continuity, the applied circuit may compensate for a certain ppm little by little (adaptive ppm compensation).

In the second step, estimating the offset generated by the audio digital signal processor (DSP) based on the error between the input and output of the audio buffer generates underrun and overrun of the audio buffer by the error between the input and output of the audio buffer to do; and converting a buffer emptying rate into an error rate based on the underrun and overrun, and generating the second offset tracking value based on the converted error rate.

5 is a diagram illustrating a detailed process of the second step.

Through the above-described first step, the audio synchronization of the SL is matched by adjusting the first SRC of the SL input terminal.

In this process, the accurately estimated input (48ksps in Fig. 5) creates an under/overrun of the Audio 1 buffer and Audio 2 buffer as much as the error of the output (48ksps + x2' offset). -x2' offset is compensated by the second SRC provided at the output terminal of the SL.

That is, the present invention generally removes the unidirectionally estimated offset, not the bidirectional offset removal method in which the offset estimated from the audio DSP is applied to the SRC. Accordingly, offset tracking is possible using simple buffer over/under buffer change and increase/decrease in the number of input/output for absolute time. It has the effect of being able to operate independently by removing it.

In addition, the sampling rate converter may be provided as many as the number of inputs or outputs. Preferably, the first sampling rate converter includes as many analog signals as the input number and the second sampling rate converter includes one. In this case, it is possible to minimize DMIPS (Dhrystone Million Instructions Per Second) usage of the SRC circuit within the SoC.

Meanwhile, the SL module calculates the delay time between the heterogeneous audio signals by using a multi-step operation method in different time sections for seamless connection between the heterogeneous audio signals.

For example, when the first tuner module of the radio reception device is currently outputting the DAB audio signal, the delay time between the DAB signal and the FM signal and between the DAB signal and the DRM FM signal is calculated in advance in the background, so that when the actual audio signal is converted It can be connected seamlessly without interruption.

Hereinafter, the seamless connection method between DAB to RadioDNS will be described in detail for convenience of explanation, but the present invention is not limited thereto. The same method can be applied to various heterogeneous audio broadcast receiving devices such as AM, IBOC to FM/AM, and the like.

In particular, the multi-step delay calculation method for seamless connection according to the present invention first calculates a rough rough Delay in a large range, and applies a small range of detailed calculations based on this. It is a method of calculating a small range based on the previously calculated rough delay even if n is changed.

Specifically, the multi-step delay calculation method for seamless connection according to an embodiment of the present invention includes a first delay time calculation step, a second delay time calculation step, and a delay time tracking step.

In the first delay time calculation step, sampling data for each of the DAB audio signal and the RadioDNS audio signal is collected for each first sampling period in a first time period, and the first extracted from the DAB audio signal among a plurality of the sampling data The first delay time is estimated by calculating a cross-correlation between the sampled data and the second sampled data extracted from the RadioDNS audio signal.

One-step calculation can be considered to require more computation than the conventional method of calculating only once. However, in the existing method of calculating only once, in order to increase accuracy, cross-correlation between DAB and RadioDNS is performed using all dense samples. Calculate.

In the second delay time calculation step, the DAB audio signal and the RadioDNS audio signal are set based on the first delay time, and for every second sampling period shorter than the first sampling period in a second time period shorter than the first time period By collecting sampling data for each signal, cross-correlation between all third sampled data extracted from the DAB audio signal and all fourth sampled data extracted from the RadioDNS audio signal is calculated to determine the detailed delay time After estimating , the detailed delay time is calculated on the first delay time.

In step 2 calculation, cross-correlation is obtained with only one sample out of 10 (decimation 10), not all samples, and only a small part is densely analyzed for all samples based on the rough rough delay value obtained in step 1. Since cross-correlation is calculated, it requires equal or less computational amount than the existing method. In addition, since the calculation of the delay change, which needs to be calculated periodically thereafter, is densely calculated in a small range, it is possible to dramatically reduce the amount of memory and computation.

This multi-step delay calculation method can be applied not only to the second step as described above, but also to three or more steps by continuously narrowing the range. can be used to increase the accuracy. In addition, since the moment of calculation for one-time calculation, the content may be a moment of silence or a moment of very small volume, so repeated calculations can reduce errors. However, large delay calculation consumes a lot of calculation amount and time, so it is difficult to calculate multiple times.

For example, if it is assumed that the delay between RadioDNS and DAB is 12.7 seconds, using the multi-step delay calculation method for seamless connection according to the present invention, when calculating the first delay in the first delay time calculation step (S10), 1 second in units of 1 second A rough delay between ~20 seconds is calculated and the first delay time is estimated to be 13 seconds.

After the delay of about 13 seconds is calculated, the delay is calculated in 0.1 second units of about +-2 seconds (11-15 seconds), and the final delay time of 12.7 seconds can be calculated. After that, the delay is always calculated in 0.1 second units between 13 seconds + -2 seconds (11-15 seconds). If there is no value that fits in this interval, it goes back to the beginning and performs the calculation of a wide range again.

In some other embodiments, the RF illustrated in FIG. 2 is characterized in that both a plurality of tuner modules and a single wideband tuner module may be provided in order to receive a plurality of RF signals for each frequency channel.

In this case, the SOC may include a plurality of tuner modules and a switch for selectively connecting to one type of tuner module among a single broadband tuner module.

The SOC according to another embodiment of the present invention having the above configuration is connected to a plurality of tuner modules to collect RF signals (hereinafter, referred to as first RF signals) from the plurality of tuner modules, and then switches a switch to a single wideband tuner module to collect an RF signal (hereinafter, referred to as a second RF signal) from a single wideband tuner module.

Thereafter, the SOC may calculate a signal-to-noise ratio of the first RF signal and a signal-to-noise ratio of the second RF signal, respectively. To this end, the SOC can calculate the signal-to-noise ratio, which is the ratio of noise included in the original signal, by comparing information on the original signal transmitted from the broadcasting station with the received RF signal. decide to do

In this case, the SOC may calculate a signal-to-noise ratio for each frequency band divided in advance, and may calculate a signal-to-noise ratio for each frequency band of the first RF signal and a signal-to-noise ratio for each frequency band of the second RF signal.

The SOC may select a tuner module having more frequency bands having an excellent signal-to-noise ratio and control the switch to be connected to the selected tuner module.

Meanwhile, the SOC repeats this signal comparison process every preset time to determine which tuner module has a better RF signal according to the movement of the radio device, and dynamically determines the reception tuner of the RF signal to determine the signal quality. This can be improved.

The technology according to the present invention as described above may be implemented as an application or implemented in the form of program instructions that may be executed through various computer components and recorded in a computer-readable recording medium. The computer-readable recording medium may include program instructions, data files, data structures, etc. alone or in combination.

The program instructions recorded in the computer-readable recording medium are specially designed and configured for the present invention, and may be known and available to those skilled in the computer software field.

Examples of the computer-readable recording medium include hard disks, magnetic media such as floppy disks and magnetic tapes, optical recording media such as CD-ROMs and DVDs, and magneto-optical media such as floppy disks. media), and hardware devices specially configured to store and execute program instructions, such as ROM, RAM, flash memory, and the like.

Examples of program instructions include not only machine language codes such as those generated by a compiler, but also high-level language codes that can be executed by a computer using an interpreter or the like. The hardware device may be configured to operate as one or more software modules for carrying out the processing according to the present invention, and vice versa.

Although the above has been described with reference to the embodiments, those skilled in the art will understand that various modifications and changes can be made to the present invention within the scope without departing from the spirit and space of the present invention described in the claims below. will be able

Claims (3)

An audio synchronization method using unidirectional offset estimation, which synchronizes an analog audio output and a digital audio output using an offset tracking value generated by a digital demodulator, the audio synchronization method comprising:
The audio synchronization method using the one-way offset estimation,
In order to remove the offset generated in the RF receiver, the first sampling rate converter provided at the input terminal of the SL (Seamless Linking) synchronizes the analog audio output and the digital audio output using the first offset tracking value generated by the digital demodulator. step;
An offset predicted to be generated by an audio DSP (Digital Signal Processor) is estimated based on the error between the input and output of the audio buffer, and a second offset tracking value for removing the estimated offset is provided at the output terminal of the SL. 2 sending to a sampling rate converter; and
An audio synchronization method using one-way offset estimation, comprising correcting an audio signal based on the second offset tracking value by a second sampling rate converter provided at an output end of the SL.
delete The method of claim 1,
Estimating the offset generated by the audio DSP (Digital Signal Processor) based on the error between the input and output of the audio buffer,
generating underrun and overrun of the audio buffer by an error between the input and output of the audio buffer;
An audio synchronization method using one-way offset estimation, characterized in that a buffer emptying rate is converted into an error rate based on underrun and overrun, and the second offset tracking value is generated based on the converted error rate.

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