Wireless Audio Stream Synchronization With Adaptive Resampling
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Solution Overview
Problem
Existing wireless audio systems face challenges in maintaining low latency and synchronizing audio streams across multiple endpoints, particularly in wireless networks, where packet loss and interference affect the synchronization of media and sample clocks, leading to channel-to-channel latency issues that degrade audio quality.
Innovation Solution
The implementation of an adjustable filter mechanism to synchronize media clocks across network endpoints, using techniques such as critically damped and over-damped loop filters, along with frequency tracking and presentation time locked loops, to maintain precise timing and reduce latency.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Ease of operation
If wireless transmission is used to distribute audio signals, then ease of installation and flexibility are improved, but latency and synchronization precision deteriorate
Solution Approach 1:
The patent implements feedback mechanisms where receiving devices send timing information back to transmitting devices, enabling dynamic adjustment of transmission timing to compensate for variable wireless delays and maintain synchronization across the network
Solution Approach 2:
The system performs preliminary timing adjustments and pre-synchronization procedures before actual audio transmission begins, allowing the system to compensate for known delays and establish synchronized operation in advance
2Adaptability or versatility
If packetized digital data is transmitted over wireless networks, then adaptability and network flexibility are improved, but synchronization precision and timing accuracy deteriorate
Solution Approach 1:
The patent introduces intermediary timing synchronization messages that travel with audio data packets, providing reference timing information that receiving devices use to resynchronize their clock recovery processes and maintain precise synchronization despite network variations
Solution Approach 2:
The system dynamically adjusts timing parameters and clock recovery mechanisms based on real-time network conditions and observed synchronization errors, allowing the system to adapt to changing wireless environment while maintaining precision
3Reliability
If clock synchronization is implemented across multiple network endpoints, then audio quality is improved, but system complexity increases
Solution Approach 1:
The patent segments the clock synchronization problem into separate clock domains (transmitting device clocks, receiving device clocks, and intermediate synchronization references), allowing each segment to be managed independently with appropriate complexity only where needed
Solution Approach 2:
The system uses simplified clock copying mechanisms where receiving devices copy timing information from transmitted reference signals rather than implementing full independent synchronization algorithms, reducing computational complexity while maintaining accuracy
4Speed
If low latency is achieved through quick radio medium access, then speed is improved, but synchronization reliability deteriorates due to packet loss and interference
Solution Approach 1:
The patent implements cushioning mechanisms in the form of timing margins and buffer periods built into the synchronization protocol, providing a safety margin that absorbs the effects of packet loss and interference without compromising overall synchronization reliability
Solution Approach 2:
The system replaces traditional mechanical clock synchronization methods with software-based timing adjustment and digital signal processing techniques that can compensate for packet loss through error correction and re-synchronization algorithms
Data Source
AI summary
Illustrative embodiments provide a system and method for synchronizing wireless audio streams. An input audio data stream including data blocks is received at a transmitter device. A presentation time for each data block is determined based on a local wall clock of the transmitter device. A packetized data stream including the data blocks and presentation timestamps is transmitted to each receiver device. The packetized data stream is received at the receiver and a local wall time is estimated using timestamp information derived from a time synchronization function counter. The received audio data is resampled based on coefficients derived from differences between the local wall time and the received presentation timestamps. Playback timing is adjusted at each receiver device based on the resampled audio data to maintain inter-channel phase coherence across the one or more receiver devices.