Audio Block Repackaging for Low Latency Distribution
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Solution Overview
Problem
Existing audio distribution systems often introduce latency and jitter, causing unsynchronized audio and video playback, particularly when audio is processed and distributed across different devices.
Innovation Solution
The method involves receiving audio samples in blocks, re-packaging them into blocks with a non-integer fraction of the original number of samples, and transmitting these over fixed duration time intervals to reduce jitter, using a wireless interface like Bluetooth, which allows for synchronized audio delivery with latency as low as 100 msec.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Adaptability or versatility
If audio content is processed and distributed through one or more systems before rendering, then audio content can be distributed to multiple devices and formatted appropriately, but latency and jitter increase causing desynchronization with video
Solution Approach 1:
The audio data stream is segmented into fixed-size packets with predetermined timestamps. Each packet contains a specific number of audio samples and is transmitted in discrete time intervals, allowing the system to maintain multiple distribution paths while preserving timing information for low-latency reconstruction at the receiving end.
Solution Approach 2:
Audio packets are pre-packaged with metadata including timestamps and sequence information before transmission. This preliminary organization allows the receiving device to buffer and reconstruct audio data with minimal processing delay, reducing overall latency while maintaining distribution flexibility.
2Manufacturing precision
If audio blocks are transmitted with varying sizes to accommodate compression requirements, then audio quality is maintained, but jitter increases causing timing inconsistency
Solution Approach 1:
The system changes the parameter of block size from variable to fixed, while maintaining audio quality through intelligent packetization schemes. Audio samples are grouped into fixed-size transmission blocks that accommodate compression requirements while ensuring consistent transmission intervals, thereby reducing jitter and timing variability.
3Loss of time
If non-integer fraction block sizes are used to reduce latency, then audio synchronization with video improves, but system complexity increases
Solution Approach 1:
The system dynamically adjusts the number of audio samples per packet based on the ratio of audio to video playback rates. This dynamic adaptation allows optimal latency reduction for different content types (e.g., 2:1 ratio for video with audio commentary, 1:1 for synchronized video) while maintaining a consistent fixed-block transmission framework that limits processing complexity.
Data Source
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AI summary
A method includes receiving data in a first series of blocks each having a first number of audio samples and repackaging the data into a second series of blocks each having a second number of audio samples. The second number of audio samples is a non-integer fraction of the first number of audio samples. The method further includes transmitting the second series of blocks over a series of fixed duration time intervals, and adjusting the pay load of adjacent time intervals to reduce jitter in the transmission of the second series of blocks.