Closed Bug 930189 Opened 11 years ago Closed 11 years ago

Audio capture from getUserMedia sometimes doesn't work

Categories

(Core :: WebRTC: Audio/Video, defect)

defect
Not set
blocker

Tracking

()

RESOLVED FIXED
mozilla27

People

(Reporter: standard8, Assigned: standard8)

References

Details

(Keywords: regression)

Attachments

(1 file)

STR 1) Visit http://mozilla.github.io/webrtc-landing/gum_test.html 2) Attempt to test audio => No audio output This appears to be a regression from bug 907817. Randell has pointed out there appears to be a rounding error in cubeb_audiounit.c line 328 the line should be: *latency_ms = (latency_range.mMinimum * 1000 + params.rate-1)/ params.rate; otherwise with small values, the latency can come out as zero, which will make audio fail.
Attached patch The fix (deleted) — Splinter Review
This is the patch with the fix as suggested by Randell. On my system, the latency_range.mMinimum is 14.0, and params.rate is 16000, hence the existing code supplies latency_ms with a value of 0, whereas with the patch we get 1, and audio then works again.
Assignee: nobody → mbanner
Status: NEW → ASSIGNED
Attachment #821546 - Flags: review?(paul)
Attachment #821546 - Flags: review?(paul) → review+
Status: ASSIGNED → RESOLVED
Closed: 11 years ago
Resolution: --- → FIXED
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