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)
Core
WebRTC: Audio/Video
Tracking
()
RESOLVED
FIXED
mozilla27
People
(Reporter: standard8, Assigned: standard8)
References
Details
(Keywords: regression)
Attachments
(1 file)
(deleted),
patch
|
padenot
:
review+
|
Details | Diff | Splinter Review |
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.
Assignee | ||
Comment 1•11 years ago
|
||
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.
Updated•11 years ago
|
Attachment #821546 -
Flags: review?(paul) → review+
Assignee | ||
Comment 2•11 years ago
|
||
Target Milestone: --- → mozilla27
Comment 3•11 years ago
|
||
Status: ASSIGNED → RESOLVED
Closed: 11 years ago
Resolution: --- → FIXED
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Description
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