Audio Processing Timer Event Synchronization
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Solution Overview
Problem
Real-time audio processing applications face issues due to unsynchronized sound card interrupts, leading to problems in timing and latency, which affect the quality of audio processing and resource utilization.
Innovation Solution
The method involves generating capture and playout interrupts, using timer events to calculate buffer sizes, and adjusting thread priorities based on latency to synchronize audio encoding and decoding processes, thereby ensuring consistent timing and resource optimization.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If sound card interrupts are used to drive real-time audio processing, then audio data can be captured and processed, but timing synchronization problems occur when interrupts are not synchronized with hardware interrupts
Solution Approach 1:
The patent introduces a timer event as an intermediary mechanism between the sound card interrupt and the audio processing thread. The timer event acts as a mediator that provides precise timing control and synchronization, ensuring that audio processing occurs at the correct intervals without relying solely on interrupt timing. This resolves the synchronization problem by decoupling the interrupt-driven data capture from the timing-critical processing decisions.
Solution Approach 2:
The patent implements a feedback mechanism where the system continuously monitors buffer size and compares it against a target latency value. Based on this feedback, the thread priority is dynamically adjusted to maintain optimal timing performance. The feedback loop ensures that timing synchronization is maintained by detecting latency deviations and correcting them through priority adjustments.
2Loss of time
If thread priority is increased to reduce latency, then audio processing timing improves, but processing resources are consumed more heavily
Solution Approach 1:
The patent applies dynamics by making thread priority adjustable rather than fixed. The system dynamically changes thread priority based on real-time buffer conditions - increasing priority when latency is high and decreasing it when latency is acceptable. This dynamic adjustment allows the system to optimize timing performance only when necessary, thereby conserving processing resources during normal operation.
Solution Approach 2:
The patent changes the parameter of thread priority based on buffer size conditions. When the buffer size indicates latency is exceeding the target, the system changes the priority parameter to a higher value. This parameter change approach allows flexible resource management while maintaining audio timing requirements.
3Measurement precision
If buffer size is increased to reduce latency, then timing precision improves, but memory usage increases
Solution Approach 1:
The patent makes buffer size a dynamic parameter that adjusts based on timing requirements and system conditions. Rather than using a large fixed buffer, the system adjusts buffer size to provide sufficient timing precision only when needed, thereby optimizing memory usage. The buffer size is modulated in response to latency measurements and thread priority changes.
Data Source
AI summary
Provided are methods and systems for processing audio using a combination of interrupts and timer events. Buffer queues and timer events are configured and utilized between audio capture and encoding processes, and also between audio decoding and playout processes to deliver a fixed-rate audio packet and higher quality audio output. Thread execution priorities may be assigned to the capture and/or playout timer events, and such priorities may be automatically adjusted to conserve processing resources. By configuring timer events to drive audio encoding and decoding processes, and utilizing buffer queues to absorb variances in the timing of audio capture and playout interrupts, the methods and systems provided are able to deliver improved audio quality while conserving processing resources.


