GITNUXSOFTWARE REVIEW
Music And AudioTop 10 Best Virtual Audio Streaming Software of 2026
Compare virtual audio streaming software tools and where they fit, including Audio Hijack, BlackHole, and JACK Audio Connection Kit for setup planning.
How we ranked these tools
Vendors set the topic, but our editors set the inclusion criteria and the evaluation framework before research starts.
Agents extract product evidence with provenance, then collect written reviews and transcribe video reviews and podcasts.
Framework: Features 40% · Ease 30% · Value 30%
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Audio Hijack is the best pick for macOS prosumers who want reliable virtual routing you can tweak as a single session because its editable block-graph chains keep capture, processing, and streaming output together, whereas BlackHole fits macOS engineers who mainly need consistent virtual loopback devices for app-to-app routing tests thanks to stable multichannel channel mapping across restarts.
Editor’s top 3 picks
Three quick recommendations before you dive into the full comparison below — each one leads on a different dimension.
Audio Hijack
Session-based block graphs let capture, processing, and streaming output stay editable as one unit.
Best for: Fits when radio-style audio chains need consistent routing, processing, and streaming control in one session.
BlackHole
Editor pickVirtual audio device with multichannel channel mapping that stays consistent across app restarts during routing tests.
Best for: Fits when macOS audio engineers need reliable virtual device routing between apps for monitoring, testing, or multichannel setup.
JACK Audio Connection Kit
Editor pickA live audio routing graph with explicit port-to-port connections drives deterministic signal paths.
Best for: Fits when local routing between audio apps matters more than built-in remote streaming publishing.
Comparison Table
Audio Hijack
prosumermacOS audio capture and routing application with virtual device output and recording capabilities.
Session-based block graphs let capture, processing, and streaming output stay editable as one unit.
Audio Hijack’s core workflow builds a chain of blocks where each block defines an audio transform or an I/O endpoint, and the chain runs continuously as a named session. Capture can target specific applications and selected audio inputs, then the chain can include effects, mix adjustments, and format controls before sending audio to a streaming output. Routing and streaming configuration live in the same session graph, which reduces handoffs between routing tools and broadcasting tools.
A tradeoff is that complex multi-destination layouts and fine-grained automation can require careful session organization instead of a central routing matrix. Audio Hijack fits situations where a broadcast chain must be edited and auditioned quickly, such as daily show production with consistent levels and repeatable processing.
- +Block-based session chains combine capture, processing, and streaming in one graph
- +Application-targeted capture enables show-specific audio routing without system-wide capture
- +Format controls support predictable encoder input for consistent broadcast results
- +Saveable session setups make repeat runs predictable across recording and streaming
- –Multi-destination streaming graphs take manual planning for clarity
- –Advanced automation depends on session organization rather than centralized scheduling
- –Live monitoring and troubleshooting can require attention to buffer and level settings
Podcast producers
App-audio capture into live stream
Consistent mix across episodes
Live radio operators
Repeatable broadcast processing chain
Fewer setup mistakes
Show 1 more scenario
Audio engineers
Audition processing before publish
Shorter iteration cycles
Run the same processing blocks for monitoring and streaming outputs to validate settings quickly.
Best for: Fits when radio-style audio chains need consistent routing, processing, and streaming control in one session.
BlackHole
open-sourceOpen-source virtual audio driver for macOS that creates a virtual audio loopback device.
Virtual audio device with multichannel channel mapping that stays consistent across app restarts during routing tests.
BlackHole from existential.audio provides a macOS virtual audio device that routes audio between apps without extra hardware. It focuses on predictable channel handling for studio workflows and supports both multichannel device layouts and low-friction WASAPI-style loop behavior on systems that can host loopback-style routing.
It also includes a small set of configuration options to pick the active channel map and drive compatibility with common audio session capture and routing patterns. The result is a dependable inter-process audio pipe for mixing, monitoring, and testing chains across multiple applications.
- +Stable macOS virtual audio device behavior for inter-app routing
- +Configurable channel mapping for multichannel studio workflows
- +Minimal latency for monitoring and repeatable audio tests
- +Works cleanly with common DAW and capture app device selection
- –macOS-only deployment limits cross-platform virtual device testing
- –Channel map changes require careful re-selection in each app
- –Does not provide a built-in network streaming or multiplexing layer
- –Limited administration features compared with enterprise streaming stacks
desktop producers and mixers
route multichannel stems between apps
repeatable channel-aware signal paths
streamers and creators
feed DAW output into capture apps
captured program audio source
Show 2 more scenarios
remote meeting hosts
send app output into conferencing
call-ready virtual audio input
A selected virtual audio source lets conferencing software receive a chosen program or processed output.
audio QA and ops teams
verify playback to conferencing routing
consistent routing verification
Stable virtual device mappings create repeatable source-to-destination paths for routing checks.
Best for: Fits when macOS audio engineers need reliable virtual device routing between apps for monitoring, testing, or multichannel setup.
JACK Audio Connection Kit
open-sourceCross-platform low-latency audio server that creates virtual connections between audio applications and hardware.
A live audio routing graph with explicit port-to-port connections drives deterministic signal paths.
JACK Audio Connection Kit uses a single audio server process that exposes audio ports for JACK-aware applications and drivers, which enables explicit channel mapping and connection control. The routing model lets users connect and disconnect specific ports without restarting applications, which reduces friction when testing different signal paths. Performance tuning uses buffer size configuration and real-time scheduling options so audio throughput stays stable during editing and monitoring.
A tradeoff is that JACK is not a full streaming encoder and publishing stack, so remote distribution and internet streaming require separate software. It fits best when a studio needs local routing between DAWs, VST plugin hosts, and custom capture tools before sending audio to a separate streaming pipeline.
- +Central routing graph supports precise port and channel connections
- +Low-latency buffer control improves monitoring stability
- +Dynamic connect and disconnect reduces need to restart clients
- +Qjackctl enables repeatable session start and device selection
- –Not an end-to-end streaming encoder and publishing solution
- –Only JACK-aware apps get native, lossless routing control
- –Sample rate and timing mismatches can force manual configuration
- –Setup complexity increases for mixed driver and app environments
Post-production editors
Patch DAW playback into monitoring chains
Faster auditioning across signal chains
Live sound engineers
Coordinate VST host and audio capture
Consistent monitoring during rehearsals
Show 1 more scenario
Custom audio tool developers
Integrate bespoke JACK client processing
Repeatable testing in shared sessions
A JACK client can register ports and exchange audio with existing desktop apps through the same server.
Best for: Fits when local routing between audio apps matters more than built-in remote streaming publishing.
Virtual Audio Cable
prosumerOriginal Windows virtual audio device driver for routing audio streams between applications.
Per-cable channel mapping and virtual endpoint registration enable precise multichannel routing on a single host.
Virtual Audio Cable creates virtual audio devices that accept one stream and expose it as a selectable input or output on the local machine. It supports WASAPI-compatible looping, WDM-style virtual endpoint routing, and per-cable channel mapping so multichannel workflows can be wired for capture and playback.
- +Creates local virtual input and output endpoints for any audio-capable application
- +Channel mapping lets multichannel routing match application channel order
- +WASAPI-compatible behavior supports modern capture paths without extra bridges
- +Low overhead driver approach suits local repeaters for audio engineering tests
- –No built-in network publishing, so streaming requires separate encoder and server tools
- –Buffer size and exclusive-mode settings can require careful tuning to avoid glitches
- –Per-cable setup relies on manual configuration rather than automated provisioning
- –Debugging routing mistakes needs audio device inspection across multiple apps
broadcast and streaming operators
loop app audio into capture software
selectable routed audio
multichannel audio workflow teams
map channels for capture and playback
configured multichannel routing
Show 2 more scenarios
system integrators building pipelines
separate capture from encoding and serving
modular capture pipeline
Virtual Audio Cable acts as the routing leg before a separate encoder and server handles distribution.
local audio testing teams
WASAPI looping for repeatable tests
repeatable test signals
WASAPI-compatible looping supports deterministic local routing for verification of playback and recording paths.
Best for: Fits when local app-to-app audio routing is needed before an external streamer.
Elgato Wave Link
creator economyVirtual audio mixing software that routes multiple audio sources into separate channels for streaming.
The Wave Link mixer combines per-source control and monitoring into a single routing workspace for live program mixes.
Wave Link provides a multi-channel mixer for microphone and desktop audio sources, with per-channel gain control and processing knobs geared toward speech and content creation workflows. Routing is handled through virtual audio endpoints, letting the Wave Link output appear as an input device in streaming software and audio apps. The mixer view makes it straightforward to manage who is speaking and how each source lands in the program mix.
- +Channel-by-channel mixer view for microphones and desktop sources
- +Monitoring and routing are configured in one place
- +Virtual output appears as an input endpoint in streaming software
- +Fast level adjustments keep program mix consistent
- –Limited automation surface for provisioning pipelines or scripted changes
- –Mix changes are operator-driven rather than workflow-first
- –Advanced multi-app routing needs manual verification per destination
Live stream audio engineers
Maintain consistent interview program mix
Stable levels during sessions
Remote podcast producers
Balance guests and host audio
Reduced post-production mix work
Show 1 more scenario
Content creators
Route game audio and voice
Cleaner on-air mix
Manage voice and desktop capture in the Wave Link mixer and feed the result into broadcasting software.
Best for: Fits when one operator needs quick, consistent multi-source mixing for streaming and recording workflows.
SteelSeries Sonar
creator economyVirtual audio mixer that routes and balances game, chat, and media audio for Windows users.
Sonar’s application-aware capture plus real-time mic processing targets streamer workflows without external routing layers.
SteelSeries Sonar targets streamers and gamers who need per-application audio routing using a mix of virtual endpoints and audio processing. It captures desktop audio and microphone sources, applies noise reduction and EQ, and then routes the processed signals to selectable outputs for OBS and other apps.
- +Per-application capture routing reduces manual audio juggling in streaming setups
- +Built-in mic processing and EQ saves setup time versus standalone tools
- +Clear monitoring controls help verify what OBS will receive before going live
- +Works within a local workstation workflow that keeps routing latency predictable
- –Windows-only routing limits use for cross-platform production pipelines
- –Virtual device selection can become confusing when multiple apps target the same endpoint
- –Limited interoperability with non-Windows audio stacks used by some broadcast tools
- –Audio processing settings require careful testing for each scene and app
streamers and recording creators
separate voice from system audio
cleaner separated recordings
streamers and live creators
reduce mic noise during broadcasts
clearer live microphone audio
Show 2 more scenarios
PC gamers
balance game chat and mic levels
tighter in-game communication
Independent level control across game, chat, microphone, and media channels supports focused listening while streaming.
podcasters and content creators
pre-process speech for recording
more consistent speech input
A leveled and equalized microphone feed routes in real time into the recording workflow.
Best for: Fits when a single Windows workstation needs per-app desktop audio routing and live mic processing.
NVIDIA Broadcast
creator economyApplication that transforms any room into a home studio with virtual microphone and speaker noise removal.
Real-time noise removal and room echo reduction running inside NVIDIA Broadcast with per-input processing controls.
NVIDIA Broadcast provides desktop and microphone audio capture, then applies GPU-accelerated processing such as noise removal and echo reduction.
- +GPU-accelerated mic cleanup with real-time noise and echo reduction
- +Works with standard Windows audio device selection for capture and output
- +Includes per-source controls for input gain and processing intensity
- +Stable virtual audio routing for stream and conferencing use cases
- –Audio processing is tied to the Broadcast app, not a chainable plugin
- –Virtual device selection requires careful Windows output and input pairing
- –Advanced routing scenarios need extra tools outside Broadcast
- –Not designed for bit-perfect passthrough or deterministic low-latency routing
streamers and broadcasters
process microphone and webcam audio
cleaner broadcast output
live production operators
route processed feeds into capture tools
enhanced signals in workflow
Show 2 more scenarios
call participants using compatible apps
apply live microphone effects for calls
processed microphone stream
Microphone effects provide a processed voice feed for compatible apps without building an external effect chain.
presenters and webcam users
deliver a changed background on camera
ready-to-stream camera output
A virtual camera feed supports background removal or modification without requiring a physical green screen setup.
Best for: Fits when streamers need real-time mic cleanup and echo control with Windows audio device selection.
AudioRelay
SMBCross-platform application that streams audio between devices over local network or internet connections.
Per-application source selection combined with explicit channel mapping for outgoing streams.
AudioRelay provides virtual audio streaming with an emphasis on routing system audio to remote or external capture endpoints. It supports per-application capture and channel mapping so engineers can control which sources are sent into a stream.
- +Application-specific audio capture reduces unwanted spillover
- +Channel mapping controls multichannel routing into the outgoing stream
- +Configurable routing targets help standardize multi-station setups
- +Operational UI supports monitoring stream state during changes
- –Advanced latency and buffer tuning requires careful setup discipline
- –Format and drift behavior varies with host audio graph and endpoints
- –Complex multi-source routing can be slower to reconfigure live
- –Extensibility depends on the supported integration surface for endpoints
remote audio engineers
send system audio to external endpoints
available remote audio stream
broadcast audio teams
capture per-application audio channels
source-targeted audio routing
Show 2 more scenarios
studio ops coordinators
run repeatable multi-endpoint configs
consistent endpoint behavior
Configuration-first deployment supports stable routing setups across multiple capture targets for sessions.
latency-sensitive workflow owners
tune routing formats for live output
managed live transport timing
Latency and format conversion settings reflect driver routing behavior in the operating model.
Best for: Fits when audio engineers need repeatable, app-level audio capture and channel-mapped streaming across multiple endpoints.
GroundControl
open-sourceFree virtual audio driver for macOS supporting up to 256 channels of loopback audio.
GroundControl’s configuration-first stream orchestration treats sources and destinations as managed entities for predictable reruns.
GroundControl, hosted at groundcontrol.app, focuses on managing live audio streaming workflows with room for programmatic control. It centers on defining audio sources, routing destinations, and monitoring stream health through a configuration-first approach.
- +Configuration-driven routing reduces per-session manual adjustments
- +Monitoring signals help catch failed or stalled stream publishing quickly
- +Automation hooks support consistent redeployments across environments
- +Clear separation of source definitions and output destinations
- –Advanced routing scenarios require more upfront configuration work
- –Operational visibility can lag during fast fail and recovery cycles
- –Some audio device edge cases depend on host OS capture stability
broadcast audio operators
define sources and destinations for streams
controlled routing during sessions
studio automation teams
automate routing changes across sessions
faster session configuration
Show 2 more scenarios
live production managers
deploy environment-style stream setups
more predictable live behavior
Environment-style deployment patterns enable consistent configuration across studio and broadcast workflows.
remote streaming support staff
monitor stream health from one control plane
earlier detection of failures
Stream monitoring ties session definitions to operational checks so issues surface during the live workflow.
Best for: Fits when teams need repeatable, automated live audio routing with operational monitoring across multiple streams.
Cleanfeed
SMBBrowser-based audio streaming and recording platform for remote interviews and podcasting.
Centralized session orchestration that manages remote participant audio feeds for live production workflows.
Cleanfeed is a virtual audio streaming service that routes live producer and studio audio through a centralized session workflow. It supports browser-based participants and engineers who need synchronized, low-latency feeds for remote recordings.
- +Session-based audio routing with clear participant role handling
- +Browser-first join flow reduces dependency on virtual audio driver setup
- +Consistent remote send and receive paths for interview-style production
- +Operational controls for keeping feeds stable during live takes
- –Limited fit for pipelines that require direct virtual audio device endpoints
- –Less control over low-level buffer tuning than local routing tools
- –External recording workflows can require extra configuration to align capture
- –Integration depth for custom transport and programmatic session provisioning is limited
video podcast production teams
record remote audio with participant delivery
separated podcast audio assets
audio editors and reviewers
review synchronized dialogue across takes
faster timing verification
Show 2 more scenarios
radio stations and broadcast teams
add remote guests to live broadcast
more consistent live contributions
Operator-controlled audio brings remote contributors into a live broadcast workflow with session control.
journalists and field teams
capture interviews without local recording gear
usable interview recordings
A browser-based workflow records usable interview audio when contributors cannot use local tools.
Best for: Fits when remote interview and studio recording needs centralized session control without virtual driver work.
Conclusion
After evaluating 10 virtual audio streaming software, Audio Hijack stands out as our overall top pick — it scored highest across our combined criteria of features, ease of use, and value, which is why it sits at #1 in the rankings above.
Use the comparison table and detailed reviews above to validate the fit against your own requirements before committing to a tool.
Frequently Asked Questions About Virtual Audio Streaming Software
How does Audio Hijack’s session graph differ from JACK’s routing graph for audio streaming workflows?
Which tool can route macOS audio between applications with multichannel channel mapping consistency?
When does Icecast-style network publishing fall outside a virtual audio device workflow?
How does GroundControl handle automation and repeatable reruns of live streaming routes?
Which setup best matches per-source live monitoring and mixing when one operator controls multiple audio inputs?
What breaks if a tool relies on OS-level audio device selection rather than a dedicated streaming transport layer?
How do application-specific capture and channel mapping differ between AudioRelay and SteelSeries Sonar?
When is JACK better than a virtual audio cable for deterministic desktop studio routing?
How do session control workflows differ between Cleanfeed and local virtual audio devices?
What security controls exist in the workflow when virtual audio capture runs across multiple applications?
Tools reviewed
Primary sources checked during evaluation.
Referenced in the comparison table and product reviews above.
How to Choose the Right Virtual Audio Streaming Software
We separate options most clearly by how reliably the product streams and receives audio across apps and devices. Look for strong connection stability, low end-to-end delay, and a workflow that keeps channels and levels predictable during everyday use.
Input source reach across apps
Why it matters
Input source reach across apps decides whether your target audio actually shows up in the stream. If coverage is narrow, you’ll end up with missing app audio or extra work to reroute it.
How to check
On a demo machine, try streaming audio from at least two different app types (for example, a browser tab and a media player) and confirm both appear at the receiver. Ask whether the product supports capturing system audio or only certain app windows, and test the exact workflow they describe.
End-to-end delay under load
Why it matters
End-to-end delay under load affects lip-sync, live monitoring, and real-time collaboration. Higher delay that grows over time can make the stream feel unusable even if it starts fine.
How to check
Play a short metronome or clap track and monitor it while the stream is running; switch the receiver to headphones/monitors for direct comparison. Then repeat while you add a second audio stream or increase traffic (if the product supports it) and note whether delay noticeably increases.
Channel and level handling
Why it matters
Channel and level handling determines whether stereo stays stereo and whether levels clip or pump. This matters because virtual audio streaming often fails “audibly” through balance errors, unexpected downmixing, or unstable gain.
How to check
Use a test audio file with known left/right tones (or a stereo sine sweep) and confirm both sides arrive correctly at the receiver. In the product’s controls, adjust output volume and watch for clipping indicators and consistent loudness between source and stream.
Clock stability during long sessions
Why it matters
Clock stability during long sessions prevents drift that causes gradual pitch change or audio dropout. This is a common difference between tools that look fine for a minute and tools that stay stable for an hour.
How to check
Run a 30–60 minute playback loop while recording the received stream to a file. Listen for gradual pitch or artifacts, and if the product offers stats, capture drift/underrun/drop counters during the run.
Automation and programmable control
Why it matters
Automation and programmable control reduces manual setup when you run the stream repeatedly or across multiple rooms. If the product lacks an automation surface, you’ll rely on manual configuration and increase the chance of errors.
How to check
Look for an API, CLI, or configuration import/export. On a demo, ask them to switch source/receiver settings and apply a saved configuration, then verify the change takes effect without manual re-tuning.
Profiles for repeatable setups
Why it matters
Profiles for repeatable setups help you keep the same routing, device selection, and levels across days. Without profiles, even minor changes can silently break a workflow.
How to check
Ask to create two different profiles (for example, different receiver devices or different channel layouts) and switch between them. Confirm the product restores the expected devices, levels, and routing within seconds and keeps the stream running cleanly.
Red flags
- Streams start but sources randomly go silent when switching apps.
- Delay increases significantly during longer sessions or with added streams.
- Stereo collapses, levels clip, or left/right signals arrive mismatched.
Fit by situation
- If you need to stream audio from multiple everyday apps, prioritize Input source reach across apps.
- If you monitor live while recording or presenting, prioritize End-to-end delay under load.
- If you run the stream for long meetings or broadcasts, prioritize Clock stability during long sessions.
Funding and competing interests
This study was co-funded by several of the vendors assessed in it. No single vendor funded it. Funders could propose the market studied.
They had no role in selecting vendors, setting or weighting criteria, analysing data or drawing conclusions, and they received the same factual-accuracy check as every other vendor assessed.
Market definition, inclusion rules and criteria weights were fixed by an editor before research began and did not change afterwards. Editorial process →
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