Audio Signal Synchronization via Latency Buffering
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Solution Overview
Problem
Current audio transmission systems face challenges in synchronizing input and output audio signals due to inconsistent synchronization between drivers, difficulty in compensating for jitter in asynchronous sample rate conversions, and differences in audio clocks between capture and playback devices.
Innovation Solution
The method involves detecting latency between audio capture and playback threads, determining the number of virtually buffered audio samples, and synchronizing signals by initializing buffers based on a target latency, or using a closed loop controller to adjust sample rates for independent audio clocks.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If audio capture and playback devices operate on the same audio clock, then synchronization between input and output signals is simplified, but inconsistent synchronization between input drivers and output drivers still occurs
Solution Approach 1:
The patent introduces a buffer as an intermediary between the capture and playback drivers. The buffer decouples the two drivers, allowing them to operate independently while maintaining synchronization. The buffer absorbs timing variations and ensures consistent data flow between capture and playback paths, resolving the inconsistency issue without requiring complex driver coordination.
Solution Approach 2:
The system performs preliminary latency detection and buffer initialization before actual audio processing begins. By detecting the latency between capture and playback threads in advance and pre-configuring the buffer accordingly, the system establishes consistent synchronization from the start, preventing driver inconsistency issues during operation.
2Adaptability or versatility
If asynchronous sample rate conversion is used to handle different audio clocks, then flexibility in handling different clock rates is improved, but jitter compensation becomes difficult
Solution Approach 1:
The patent implements a feedback mechanism where the system continuously monitors the latency between capture and playback threads and adjusts the buffer configuration accordingly. This closed-loop approach compensates for jitter by detecting timing variations and correcting them through dynamic buffer management, maintaining reliable synchronization even with asynchronous sample rate conversion.
Solution Approach 2:
The system dynamically adjusts buffer parameters based on detected latency and timing conditions. Rather than using fixed buffer configurations, the system adapts buffer sizes and initialization parameters in response to varying clock rates and timing conditions, enabling effective jitter compensation while maintaining flexibility in handling different audio clocks.
3Measurement precision
If buffer is initialized with physically buffered samples, then synchronization accuracy is improved, but the complexity of determining optimal buffer size increases
Solution Approach 1:
The system performs self-measurement of latency between capture and playback threads and uses this measured latency to automatically determine the appropriate buffer initialization parameters. The system serves itself by detecting its own timing characteristics and configuring the buffer accordingly, achieving high synchronization accuracy without requiring external calibration or complex manual configuration.
Data Source
AI summary
Techniques related to input and output signal synchronization and latency jitter compensation for audio systems are discussed. Such techniques may include determining a number of virtually buffered samples based on a detected latency between an audio capture thread and an audio playback thread and synchronizing an audio input signal and an audio output signal based on the number of virtually buffered samples.