Real-Time Audio Streaming Packet Buffering for Latency Reduction
Find Innovative SolutionsGenerate Solutions
Solution Overview
Problem
Existing audio streaming systems experience significant latency and buffer requirements when transmitting live audio streams to mobile devices, especially in environments with varying network conditions, which can lead to desynchronized audio and video experiences.
Innovation Solution
The system generates discrete audio data packets from a live audio stream and transmits these packets in real-time to mobile computing devices, using a buffer to store each packet and generate copies for multiple devices, thereby minimizing latency and buffer requirements.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If large data buffers are used to reassemble packets and remove timing jitter, then audio quality is preserved, but latency increases significantly
Solution Approach 1:
The system performs preliminary actions by pre-ordering packets based on sequence numbers before actual playback is needed. The client device receives and caches packets in advance, organizing them in the correct sequence so that when playback time arrives, the audio data is already ready to be played immediately without requiring large buffers for reassembly during playback.
2Adaptability or versatility
If packets are transmitted over the internet with varying network conditions, then routing flexibility is achieved, but packet timing irregularity and out-of-order arrival occur
Solution Approach 1:
The system implements feedback mechanisms where the client device monitors the arrival of packets and their sequence numbers. When packets arrive out of order or with timing irregularities, the client uses the sequence number information to detect these issues and compensates by waiting for missing packets or reordering them correctly before playback, thereby maintaining stable and synchronized audio output despite network variability.
3Reliability
If live radio programs are streamed with several seconds of latency, then network buffering is achieved, but real-time listening experience is degraded
Solution Approach 1:
The system performs preliminary actions by pre-ordering and pre-caching audio packets at the client device before they are actually needed for playback. By organizing packets in the correct sequence in advance and having them ready in the device's memory, the system eliminates the need for several seconds of buffering latency, enabling real-time listening experience where audio plays back immediately upon arrival without significant delay.
Data Source
AI summary
Systems and methods for capturing and distributing a live audio stream of a live event in real-time to a plurality of mobile computing devices are described. The system comprises at least one memory configured to store computer-executable instructions and processing device(s) coupled to the at least one memory and at least one audio input device configured to receive the live audio stream. Upon receipt of the live audio stream, the processing device(s) generate, in real-time, a plurality of discrete audio data packets from the live audio stream, and transmit that respective discrete audio data packet over a first network for receipt by the mobile computing devices. In respect of each one of the discrete audio data packets, a copy of the discrete audio data packet is transmitted for receipt by each one of the mobile computing devices prior to transmitting another one of the discrete audio data packets.