Fuzzes the config and audio inputs to GainControlImpl.
Seems able to cover a few hundred lines of code that the APM fuzzer hasn't been able to reach.
Bug: webrtc:9413
Change-Id: I32776505be9c416ec03113c12437a92dcfadd827
Reviewed-on: https://webrtc-review.googlesource.com/84589
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23709}
We found out that
int16_t x = test::FuzzDataHelper::ReadOrDefaultValue(0)
reads 4 bytes from the fuzzer input instead of 2. That means that
almost half the bits in the input data to audio_processing_fuzzer are
ignored. This change adds template arguments to force reading 2 bytes
when we only need 2.
We also add a small manually generated corpus. During local testing we
let the fuzzer run for a few hours on an empty corpus. Adding the
manually-generated files resulted in an immediate coverage increase by
~3%, and then by another 3% over the next few hours.
The manually generated corpus contains a short segment of speech with
real echo. We suspect that triggering Voice Activity Detection or echo
estimation filter convergence can be difficult for an automatic
fuzzer.
We remove the Level Controller config. We read 20 bytes extra after the
config to guard against future configuration changes.
Bug: webrtc:7820
Change-Id: If60c04f53b27c519c349a40bd13664eef7999368
Reviewed-on: https://webrtc-review.googlesource.com/58744
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22269}
Rtp Packets in webrtc expected to be less that 1500,
i.e. way less that 2^16 bytes for extensions block.
This CL explicitly discards longer extension.
Bug: chromium:809046
Change-Id: Ibed33b51bafc3fd4804ec135f66110c6d2796734
Reviewed-on: https://webrtc-review.googlesource.com/48061
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21910}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
