Adaptive Sampling Rate Conversion for Asynchronous Audio
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Solution Overview
Problem
Real-time audio transmission over asynchronous transmission mediums faces challenges due to fluctuations in sampling rates between transmitter and receiver, leading to signal distortions and interruptions, especially in sonar technology where continuous audio signals are critical.
Innovation Solution
Adaptive sampling rate conversion using a time-variant low-pass filter, implemented with a digital FIR filter, dynamically adjusts the sampling rate ratio by monitoring buffer fill levels and history, ensuring synchronization between input and output data streams despite independent clock sources.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If a separate proprietary digital transmission path is used with synchronized clock rates, then audio signal transmission quality is improved, but device complexity and cost increase due to special modules
Solution Approach 1:
The patent introduces a jitter buffer as an intermediary component between the network transmission path and the audio playback device. This buffer absorbs timing variations and sampling rate fluctuations, mediating between the asynchronous network clock and the fixed playback clock, thereby maintaining audio quality without requiring proprietary synchronized transmission paths
Solution Approach 2:
The patent dynamically adjusts the jitter buffer delay parameter based on network conditions and sampling rate variations. By changing the buffer delay parameter adaptively, the system maintains optimal audio transmission quality while using standard asynchronous transmission mediums, avoiding the need for fixed proprietary configurations
2Device complexity
If standardized transmission mediums like Ethernet are used for asynchronous transmission, then device complexity is reduced, but signal dropouts and crackling noises occur due to sampling rate fluctuations
Solution Approach 1:
The patent implements a dynamic jitter buffer that continuously adapts its delay characteristics in response to varying sampling rates and network jitter. This dynamic adjustment prevents buffer underflow and overflow conditions that cause signal dropouts and crackling, while maintaining compatibility with standardized asynchronous transmission mediums
Solution Approach 2:
The patent employs feedback mechanisms that monitor the fill level of the jitter buffer and adjust the delay parameter accordingly. This closed-loop control ensures continuous audio signal transmission by preventing buffer emptying (which causes dropouts) and overflow (which causes distortion), thereby maintaining signal reliability with standard transmission mediums
3Reliability
If network jitter is reduced by using a jitter buffer, then audio transmission quality is improved, but additional delay is introduced
Solution Approach 1:
The patent implements a dynamic jitter buffer that continuously adapts its delay characteristics in response to varying sampling rates and network jitter. This dynamic adjustment prevents buffer underflow and overflow conditions that cause signal dropouts and crackling, while maintaining compatibility with standardized asynchronous transmission mediums
Solution Approach 2:
The patent employs feedback mechanisms that monitor the fill level of the jitter buffer and adjust the delay parameter accordingly. This closed-loop control ensures continuous audio signal transmission by preventing buffer emptying (which causes dropouts) and overflow (which causes distortion), thereby maintaining signal reliability with standard transmission mediums
4Adaptability or versatility
If independent clock generators are used at transmitter and receiver, then adaptability is improved, but sampling rate drift causes signal distortions and interruptions
Solution Approach 1:
The patent introduces a jitter buffer as an intermediary component between the network transmission path and the audio playback device. This buffer absorbs timing variations and sampling rate fluctuations, mediating between the asynchronous network clock and the fixed playback clock, thereby maintaining audio quality without requiring proprietary synchronized transmission paths
Solution Approach 2:
The patent dynamically adjusts the jitter buffer delay parameter based on network conditions and sampling rate variations. By changing the buffer delay parameter adaptively, the system maintains optimal audio transmission quality while using standard asynchronous transmission mediums, avoiding the need for fixed proprietary configurations
Data Source
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AI summary
The invention relates to a method for compensating for fluctuations of scan rates f A , f DAC , which are based on an input data stream 11 connected to a receiver 6 and an output data stream 12 output by the receiver 6. The input data stream 11 was first transferred via a network jitter causing communication network 5 and then intermediately stored on the receiver 6 in a buffer 20, the level thereof being monitored by a buffer read indicator 32 and kept in the region of a target value. In order to enable transmission of data signals, in particular audio data signals, without a fixed scan rate ratio being present between the data signals of the transmitter 2 and the receiver 6, an adaptive scan rate conversion 28 is proposed according to the invention. This occurs by means of a digital filter 26 having dynamically variable filter coefficients, wherein the filter coefficients are calculated depending on a non-integer portion 36 of the buffer read indicator 32. In addition, an estimated value of the scan rate ratio is determined by means of the integer portion 24 of the buffer read indicator 32. The invention further relates to a device for executing a method of said type.