How system audio reaches the client, the opt-in AAC compression path, and the minimize/refocus behavior.
System audio rides over the RDPSND virtual channel as 16-bit stereo PCM at 44.1 kHz. ScreenCaptureKit only supports 8 / 16 / 24 / 48 kHz, so the capture loop captures at 48 kHz and resamples to 44.1 with rubato before sending. The 44.1 kHz output is empirically load-bearing — the earlier 48 kHz feed over-fed mstsc into multi-second backlogs, and 44.1 fixed it (the precise mechanism is a pacing/rate-accounting effect; Windows resamples shared-mode streams to the endpoint mix format either way). A generation counter on the audio factory keeps a client reconnect from leaving a second capture loop feeding the channel. The vendored ironrdp-server carries a single patch that makes dispatch_server_events keep the newest queued waves on per-batch overflow instead of the oldest — without it, a one-off video-encode stall would bake a permanent audio-latency offset into the session.
The capture loop also self-heals a dead SCK audio stream: over a long session ScreenCaptureKit can stop delivering samples or transiently fail to start, which previously left the connection silent for the rest of the session (video is a separate stream and kept running). The loop now rebuilds the audio SCStream with capped exponential backoff (250 ms → 5 s) on both start failures and mid-stream end, resetting the backoff once a sample arrives; the generation guard still retires it on reconnect, so there's no double-capture.
AAC compression (opt-in, --enable-aac). By default audio is uncompressed PCM (~1.4 Mbit/s). Pass --enable-aac to encode it as AAC-LC over RDPSND (WAVE_FORMAT_AAC_MS, ~128 kbps by default — about 11x smaller), which matters over WAN or constrained links. The encoder is AudioToolbox (software AAC-LC); the wire payload is raw AAC access units. The server advertises AAC ahead of PCM, so clients that decode it (mstsc, Microsoft Remote Desktop / Windows App, FreeRDP built with AAC support) negotiate AAC automatically while clients without it fall back to PCM transparently. It's off by default because AAC adds ~40–50 ms of encoder priming latency — on a LAN, PCM's zero added latency is the better default. Tune the bitrate with --aac-bitrate (default 128000; 96000 saves the most bandwidth, 192000 is near-transparent for music).
Mute on minimize (default-on, opt out with --no-mute-on-minimize). When the client minimizes its window it sends the standard SuppressOutput { None } PDU; the server stops emitting both EGFX video frames and RDPSND waves until the client refocuses (RefreshRectangle / SuppressOutput { Some(rect) }). Without this, mstsc accumulates a backlog of video frames + audio waves during a long minimize that has to chew through on refocus, producing several seconds of input lockout, audio drift, and a video catch-up storm. With it, you get a brief audio gap on refocus and audio + video resume in sync. Both gates are debounced (1 s) so transient SuppressOutput flaps mstsc emits under wire pressure (e.g., during a heavy local cargo build) don't oscillate the mute and cause stutter. Pass --no-mute-on-minimize if you specifically want audio to keep playing while the client window is minimized — accepting that audio will drift by however long was spent minimized.
The other direction: the client's microphone becomes a real input device on
the Mac, "macrdp Microphone", which Zoom, FaceTime, QuickTime and Teams can
record from. Off by default; turn it on with --enable-microphone-redirection
(config ENABLE_MICROPHONE_REDIRECTION=1, or the controller's Settings →
Redirection → Microphone toggle).
Setup, once: install the "macrdp Microphone" driver from the menu-bar
controller — Settings → Redirection → Microphone → Install macrdp
Microphone…. It asks for an administrator password and restarts Core Audio,
which briefly interrupts sound. The same place shows when a newer macrdp bundles
a newer driver (Update…) and removes it (Remove…). Without a controller,
run /Applications/macrdp.app/Contents/Resources/install-audio-plugin.sh.
On the client: mstsc → Local Resources → Remote audio → Settings →
Record from this computer, set before connecting. FreeRDP: /microphone.
What to expect: the device carries the client's audio with about 100 ms of latency, and reads as silence when no session is streaming. After a network stall it skips ahead to stay current rather than playing late audio, which can cause a brief click.
Limits:
- Other people's accounts on this Mac can listen in while a session is
streaming. They can't inject audio, and nothing is kept after the session ends.
On a Mac with other user accounts, treat an active redirected mic as audible to
them. (Why:
docs/macos-gotchas.md, item 5.) - 44.1 kHz only for now (16-bit PCM). mstsc offers it. A client that can't gets no mic, and macrdp logs why.
Debugging: MIC_DUMP=1 in config.env (env MACRDP_MIC_DUMP=1) writes
the received audio to a WAV file under $TMPDIR instead of feeding the device —
play it back to check what the client actually sent.