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7b201012bc7f7cd95249b8314d0d7ebabe966d8b
platform-external-webrtc/modules/audio_coding
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Patrik Höglund 7b201012bc Flip histograms to true by default, fix unit in isac_fix_test. 
Requires downstream changes for all WebRTC perf tests, and
a corresponding recipe change so isac_fix_test starts using the new
flow.

Bug: chromium:1029452
Change-Id: I8918fca9bef003d365037c1c6bf7c55747dfed99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170633
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30906}
2020-03-26 19:36:44 +00:00
..
acm2
ACM: Corrected temporary buffer size
2020-03-12 12:23:20 +00:00
audio_network_adaptor
Use newer version of TimeDelta and TimeStamp factories in modules/
2020-02-10 11:49:57 +00:00
codecs
Flip histograms to true by default, fix unit in isac_fix_test.
2020-03-26 19:36:44 +00:00
include
ACM: Corrected temporary buffer size
2020-03-12 12:23:20 +00:00
neteq
Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
2020-03-11 12:08:32 +00:00
test
ACM: Corrected temporary buffer size
2020-03-12 12:23:20 +00:00
audio_coding.gni
Don't select audio codecs depending on GN vars build_with_{chromium|mozilla}
2017-11-01 18:59:27 +00:00
BUILD.gn
Implement AudioEncoder::GetFrameLengthRange() for all audio encoders.
2020-03-25 22:19:21 +00:00
DEPS
…
OWNERS
Remove wildcard ownership for build files.
2020-02-19 14:05:46 +00:00