Cloud-Based DSP Mixing for Low-Latency Live Performance

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Solution Overview

Problem

Existing digital signal processing technologies for live performances with multiple participants face computational overload and latency issues, particularly when scaling beyond a few performers, especially over consumer-grade Internet connections.

Innovation Solution

A cloud-based streaming platform utilizing a hub server with digital signal processing (DSP) to manage audio streams, employing separate signal chains for low-latency performer mixes and high-quality audience mixes, leveraging cloud computing for scalable and efficient audio processing across distributed participants.

Engineering Contradictions & Design Principles

VSEngineering Contradiction Analysis

1Manufacturing precision

If digital signal processing is applied to create realistic live performance environments for hundreds of participants, then audio quality and realism are improved, but computational requirements become overwhelming for client devices

Engineering Contradiction:
Improveaudio qualityVSAvoidcomputational requirements
Core Design Contradiction:
Manufacturing precisionVSDevice complexity

Solution Approach 1:

A cloud-based audio processing server is introduced as an intermediary between client devices and the audio mixing process. The server receives audio streams from multiple participants, applies digital signal processing effects, and distributes the mixed audio back to participants and audiences. This eliminates the need for complex local processing on client devices while maintaining high audio quality and realistic performance environments.

Inventive Principle:
Principle #24Intermediary (Mediator)

2Manufacturing precision

If multiple DSP processing steps are applied to audio channels to enhance realism, then audio processing capability is improved, but processing latency increases

Engineering Contradiction:
Improveaudio processing capabilityVSAvoidprocessing latency
Core Design Contradiction:
Manufacturing precisionVSLoss of time

Solution Approach 1:

The audio processing system is segmented into separate processing paths with different latency characteristics. A first path processes audio for performers with minimal latency to maintain synchronization, while a second path processes audio for audiences with higher latency tolerance to apply comprehensive DSP effects. This segmentation allows each path to be optimized for its specific requirements without compromising the other.

Inventive Principle:
Principle #1Segmentation

3Adaptability or versatility

If a hub server processes audio for hundreds of participants, then scalability is improved, but network bandwidth and server resources are consumed

Engineering Contradiction:
ImprovescalabilityVSAvoidnetwork bandwidth
Core Design Contradiction:
Adaptability or versatilityVSQuantity of substance

Solution Approach 1:

Multiple individual audio streams from hundreds of participants are merged into a single mixed audio output that is distributed to all participants and audiences. Instead of requiring each participant to receive separate processed streams, the system combines all inputs at the server and distributes the mixed result, significantly reducing network bandwidth requirements while maintaining scalability.

Inventive Principle:
Principle #5Merging (Combining)

Data Source

PatentUS12411649B2Digital signal processing for cloud-based live performance
Publication Date: 2025.09.09 JACKTRIP LABS INC
  • US12411649B2 patent drawing
  • US12411649B2 patent drawing
  • US12411649B2 patent drawing

AI summary

An audio processing and streaming server system includes an input bus, a digital signal processor (DSP), a first and second stream generators. The input bus includes a plurality of input audio channels, each corresponding to a respective input stream of a plurality of input streams, each received at the server from one of a plurality of audio interface devices. The DSP is configured to apply at least one signal chain to at least some of the plurality of input audio channels; and output first and second master audio mixes in accordance with the at least one signal chain. The first stream generator is configured to convey the first master audio mix to the audio interface devices in a first live stream with no added buffering latency, and the second stream generator is configured to buffer and convey the second master audio mix in a second live stream.