Make the SIP audio mixer input buffer size configurable#727
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The mixer's per-input buffer defaults to media-sdk's 5 frames (100 ms), which can be too small for bursty inputs: chunks larger than the buffer overflow on arrival and underrun between them, restarting the mixer and injecting silence. Add a `mixer_input_buffer_frames` config option that passes `mixer.WithInputBufferFrames(n)` to the room and DTMF mixers when n > 0. Default unchanged (0 = library default), so existing deployments are unaffected.
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Codecov Report✅ All modified and coverable lines are covered by tests. Additional details and impacted files@@ Coverage Diff @@
## main #727 +/- ##
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+ Coverage 65.25% 66.36% +1.11%
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Lines 6588 7534 +946
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- Misses 1915 2080 +165
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Summary
Adds a
mixer_input_buffer_framesconfig option to size the SIP mixer's per-input buffer. Default unchanged (media-sdk's 5 frames / 100 ms) — opt-in, existing deployments unaffected.Why
The bridge mixes each input through media-sdk's mixer, whose input buffer defaults to 5 frames (100 ms). That can be too small for bursty sources: chunks longer than the buffer overflow on arrival and underrun between them, so the mixer restarts and emits silence — choppy outbound audio.
mixer.WithInputBufferFramesalready exists in media-sdk; the bridge just never passed it.Driving the real mixer offline with identical bursty input, only the buffer changed:
Change
mixer_input_buffer_frames(int;0= library default).mixer.WithInputBufferFrames(n)to the room and DTMF mixers only whenn > 0.Trade-off
A deeper buffer adds up to
n × 20 msof per-input latency (15 frames raises the rebuffer threshold from 60 ms to 160 ms), so the default stays at 5 and operators raise it only when they hit choppy realtime-over-SIP audio. The same 5→15 value is hardcoded in a fork (Bigly-Sales#6); this keeps it configurable instead. Related: livekit/media-sdk#38 tuned mixer timing but left the input buffer at 5.