Audio Playback Device Buffer Synchronization via Frequency Adjustment
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Solution Overview
Problem
Existing audio playing devices face synchronization issues due to temperature-induced zero drift, causing changes in the oscillation signal frequency, which can lead to mismatch between the first and second sampling frequencies, resulting in unsynchronized electronic and playing devices and affecting audio playback quality.
Innovation Solution
A method that detects the number of audio sampling points in a buffer area and adjusts the frequency-division parameter to determine a second sampling frequency, ensuring synchronization between the electronic and playing devices by adjusting the sampling frequency based on predefined thresholds or ratios to maintain an optimal quantity range of audio sampling points.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If the oscillation signal frequency is kept fixed to ensure synchronization between electronic device and playing device, then the sampling frequency remains stable, but temperature-induced zero drift causes frequency changes and desynchronization
Solution Approach 1:
The patent implements a feedback mechanism by continuously monitoring the number of audio sampling points in the buffer area and using this information to dynamically adjust the frequency-division parameter. When the buffer quantity deviates from the optimal range, the system adjusts the frequency-division parameter to correct the sampling frequency mismatch, thereby maintaining synchronization between the electronic device and playing device despite temperature-induced zero drift.
Solution Approach 2:
The patent changes the frequency-division parameter dynamically based on buffer area monitoring. By adjusting this parameter, the system modifies the second sampling frequency to match the first sampling frequency from the electronic device, compensating for frequency drift caused by temperature changes and maintaining reliable synchronization.
2Reliability
If the frequency-division parameter is adjusted dynamically to compensate for zero drift, then synchronization is maintained, but the device complexity increases
Solution Approach 1:
The system performs self-adjustment by automatically monitoring its own buffer area status and correcting frequency mismatches without external intervention. The playing device independently detects buffer quantity deviations and adjusts its frequency-division parameter accordingly, eliminating the need for complex external control mechanisms while maintaining high synchronization accuracy.
Solution Approach 2:
The simple feedback loop monitors buffer quantity and directly controls frequency-division parameter adjustment. This closed-loop control mechanism achieves reliable synchronization with minimal added complexity by using the buffer status as both the sensor and actuator control signal.
3Manufacturing precision
If resampling algorithms are used to correct frequency mismatch, then audio playback quality is maintained, but computational complexity and processing time increase
Solution Approach 1:
The patent replaces complex software-based resampling algorithms with a simpler hardware-level frequency adjustment mechanism. By modifying the frequency-division parameter, the system directly adjusts the sampling frequency to match between devices, avoiding the need for computationally intensive resampling operations while maintaining audio playback quality.
Solution Approach 2:
Instead of processing audio data through complex resampling algorithms, the system changes the frequency-division parameter to prevent frequency mismatch at the source. This proactive parameter adjustment maintains audio quality without the computational overhead and processing delays associated with post-capture resampling.
Data Source
AI summary
The present disclosure relates to a method for playing audio, an apparatus for playing audio, a playing device, and storage medium, and belongs to the field of electronic technology. The method includes detecting an amount of audio sampling points in a buffer area that is configured to store the audio sampling points received from an electronic device based on a first sampling frequency, wherein the audio sampling points are obtained by decoding audio data from the electronic device; when a first quantity of the sampling points based on the first sampling frequency is out of a default quantity range, adjusting a preset first frequency-division parameter to obtain a second frequency-division parameter; determining a second sampling frequency based on the second frequency-division parameter; and extracting the audio sampling points from the buffer area based on the second sampling frequency for playing audio data.