Clock Skew Compensation in BLE Audio Streaming
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Solution Overview
Problem
In audio data streaming, clock skew between sender and receiver devices can cause buffer overruns or underruns, leading to audio inconsistencies and artifacts due to differing clock frequencies, which existing solutions struggle to fully address effectively.
Innovation Solution
The method involves measuring the connection interval between consecutive data exchanges in a Bluetooth Low Energy (BLE) communication system to correct the supply frequencies of both the sender and receiver devices, using the connection interval as a shared clock signal to synchronize their frequencies and prevent buffer issues through a control loop.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If clock skew compensation is implemented using sample rate conversion, then buffer overflows and underflows are prevented, but audible artifacts and pitch changes are introduced
Solution Approach 1:
The patent applies preliminary action by measuring the connection interval between consecutive data exchanges before processing the audio stream. This measurement is used to pre-calculate clock correction values that are then applied to synchronize the playback clock with the transmission clock, preventing buffer issues before they occur rather than correcting them after they manifest as artifacts
Solution Approach 2:
The patent changes the parameter of clock frequency by using the measured connection interval to calculate a corrected supply frequency. This corrected frequency is then used to clock the audio processing operations, effectively adjusting the clock parameter to compensate for skew between sender and receiver devices without introducing audible distortions
2Reliability
If complex packet duplication or dropping approaches are used, then buffer issues are avoided, but discontinuities and audio quality degradation occur
Solution Approach 1:
The patent implements feedback by continuously measuring the connection interval between data exchanges and using this information to dynamically adjust the supply frequency. This closed-loop approach ensures that the playback rate is continuously synchronized with the actual transmission rate, preventing both buffer overflows and the need for packet duplication or dropping that would cause discontinuities
3Reliability
If playback speed is dynamically changed to maintain buffer queue length, then buffer overflows and underflows are prevented, but audio timing accuracy and synchronization are compromised
Solution Approach 1:
The patent applies preliminary action by measuring the connection interval before audio processing begins. This pre-measured interval is used to calculate a corrected supply frequency that is then applied consistently throughout the audio stream, establishing accurate timing from the start rather than dynamically adjusting playback speed during the stream, which would compromise timing accuracy
Data Source
AI summary
A device includes a transceiver, which, in operation, exchanges data packets over a wireless communication link with a remote device during connection events separated by a determined connection interval. The device includes data processing circuitry, which, in operation, measures an indication of a duration of time between at least two consecutive connection events associated with the remote device and adjusts a bit rate based on the measured indication of the duration of time and the determined connection interval.