Adaptive Jitter Buffer for Voice Recording Quality
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Solution Overview
Problem
Existing jitter buffers in packet switching systems face a trade-off between voice quality and delay, resulting in quality loss due to packet loss, as they are typically too short for real-time conversations but inadequate for high-quality recordings.
Innovation Solution
Adaptive jitter buffer management that dynamically adjusts buffer size based on whether the audio is for real-time conversation or recording, using a larger buffer for recordings to maintain high quality without significant delay impacts.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Loss of time
If a short jitter buffer (10-100 milliseconds) is used for real-time conversations, then delay is minimized for acceptable human factors, but voice quality deteriorates due to packet loss from jitter
Solution Approach 1:
The system dynamically adjusts the jitter buffer size based on the application type. For real-time conversations, a short buffer (10-100 milliseconds) is used to minimize delay. For recorded voice applications, the buffer is extended to a longer duration to capture more packets and improve voice quality by reducing packet loss from jitter.
Solution Approach 2:
The invention changes the buffer length parameter according to the specific application requirements. By detecting whether the audio is for real-time processing or recording, the system selects appropriate buffer parameters: short buffers for interactive applications and extended buffers for recording applications where quality is prioritized over immediate playback.
2Reliability
If a long jitter buffer is used to improve voice quality by capturing more packets, then voice quality improves, but unacceptable delays occur during telephone conversation
Solution Approach 1:
The system dynamically adjusts the jitter buffer size based on the application type. For real-time conversations, a short buffer (10-100 milliseconds) is used to minimize delay. For recorded voice applications, the buffer is extended to a longer duration to capture more packets and improve voice quality by reducing packet loss from jitter.
Solution Approach 2:
The invention changes the buffer length parameter according to the specific application requirements. By detecting whether the audio is for real-time processing or recording, the system selects appropriate buffer parameters: short buffers for interactive applications and extended buffers for recording applications where quality is prioritized over immediate playback.
3Device complexity
If a fixed jitter buffer size is used for all applications, then device complexity is reduced, but adaptability to different application requirements (real-time vs. recording) is lost
Solution Approach 1:
The system dynamically adjusts the jitter buffer size based on the application type. For real-time conversations, a short buffer (10-100 milliseconds) is used to minimize delay. For recorded voice applications, the buffer is extended to a longer duration to capture more packets and improve voice quality by reducing packet loss from jitter.
Solution Approach 2:
The invention changes the buffer length parameter according to the specific application requirements. By detecting whether the audio is for real-time processing or recording, the system selects appropriate buffer parameters: short buffers for interactive applications and extended buffers for recording applications where quality is prioritized over immediate playback.
Data Source
AI summary
A method and apparatus control effects of packet transmission delay on recorded audio information by determining if received audio information is to be used for a real time conversation or to be recorded; using a first size of jitter buffer upon the determined use being for the real time conversation; and using a second size of jitter buffer upon the determined use being to record the audio information whereby the second size of jitter buffer is larger than the first size of jitter buffer.