Audio Signal Interleaving for Packet Loss Recovery
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Solution Overview
Problem
Existing audio data transmission methods over packet networks face significant challenges due to packet loss, which degrades speech intelligibility and audio quality, and current solutions like forward error correction increase bandwidth consumption, making them undesirable in bandwidth-limited networks.
Innovation Solution
The method involves transforming audio signals into frequency domain components, interleaving these components across multiple frames, and distributing them into packets, allowing for robust reconstruction of audio signals even with packet loss without increasing bandwidth consumption.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If forward error correction (FEC) is used to recover lost packets, then speech intelligibility and audio quality are improved, but bandwidth consumption increases significantly
Solution Approach 1:
The audio signal is transformed into frequency domain components and segmented into multiple sub-bands. These sub-bands are then distributed across different packets, allowing selective recovery of important frequency components even when some packets are lost, thereby maintaining speech intelligibility without requiring redundant data in every packet.
Solution Approach 2:
Different frequency sub-bands are assigned different priorities based on their importance for speech intelligibility. Critical low-frequency components are protected with higher redundancy, while less critical high-frequency components use lower redundancy. This local quality approach optimizes bandwidth usage by allocating protection resources where they are most needed.
2Reliability
If forward error correction (FEC) is used to recover lost packets, then audio quality is improved, but bandwidth consumption increases
Solution Approach 1:
The audio spectrum is divided into multiple frequency sub-bands that are transmitted in separate packets. This segmentation allows the system to recover audio quality by reconstructing missing sub-bands using interpolation and synthesis techniques, avoiding the need to transmit redundant copies of all audio data.
Solution Approach 2:
The system dynamically adjusts the amount of redundancy and error protection applied to different frequency sub-bands based on their perceptual importance. By changing the protection parameters selectively rather than uniformly across all audio data, the system maintains audio quality while minimizing overall bandwidth consumption.
3Quantity of substance
If receiver-only approaches are used for packet loss recovery, then bandwidth consumption is reduced, but speech intelligibility and audio quality are limited
Solution Approach 1:
The transmitter performs preliminary processing of the audio signal by transforming it into frequency sub-bands and organizing packets with strategic redundancy before transmission. This preliminary action enables the receiver to effectively recover lost packets using built-in redundancy and reconstruction algorithms, achieving good speech intelligibility without requiring excessive bandwidth for error correction data.
Data Source
AI summary
Methods and corresponding apparatuses for transmitting and receiving audio signals are described. A transformation is performed on the audio signals in units of frame in order to obtain transformed audio data of each frame, said transformed audio data consisting of multiple signal components in the frequency domain. These signal components of each frame are distributed into multiple adjacent packets in order to generate packets in which signal components distributed from multiple frames are interleaved. Subsequently, the generated packets are transmitted. Accordingly, in case that packet loss occurs during transmission, the audio signals can be recovered based on the received signal components without consuming additional bandwidth. Therefore, robustness against packet loss can be achieved with little overhead.