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Alex Loiko c7b18fef19 Shifted value doesn't fit in 'int32_t'. 
This CL replaces one 'int32_t' with 'uint32_t'. The value is a
non-negative energy, and the number of leading zeros is
computed. During computation, a shift can cause it to overflow.

Issue was found by the Audio Processing fuzzer.

Bug: chromium:778939, chromium:778921, chromium:778919
Change-Id: I3d7e0b547e6b0edcd9995903517ea851142a08c1
Reviewed-on: https://webrtc-review.googlesource.com/16433
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20470}
2017-10-28 10:22:32 +00:00
api Fix clang style warnings in api/candidate.h 2017-10-26 23:22:18 +00:00
audio New classes RefCounter and RefCountedBase. 2017-10-23 11:46:47 +00:00
build_overrides Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
call Reland "Add fine grained dropped video frames counters on sending side" 2017-10-25 09:32:15 +00:00
common_audio Removing undefined left shifts in AudioProcessing 2017-10-25 13:35:36 +00:00
common_video Use $rtc_libyuv_dir in common_video/BUILD.gn, not hard-encoded "libyuv" 2017-10-16 16:05:37 +00:00
data WebRTC: Replace ProjectRootPath by ResourcePath 2016-11-22 18:43:05 +00:00
examples Revert ObjC API changes for BWE allocation strategy 2017-10-27 12:33:24 +00:00
infra Temporarily remove linux_ubsan from commit queue 2017-10-27 09:49:17 +00:00
logging Prevent unbounded memory consumption through RtcEventLogImpl::config_history_ 2017-10-13 10:47:26 +00:00
media Remove FlexFEC feedback params from internal encoder factory 2017-10-26 11:33:27 +00:00
modules Shifted value doesn't fit in 'int32_t'. 2017-10-28 10:22:32 +00:00
ortc Disable flaky test OrtcFactoryIntegrationTest.BasicTwoWayAudioVideoRtpSendersAndReceivers. 2017-09-27 09:14:28 +00:00
p2p Move inlined methods from p2p/base/packetsocketfactory.h 2017-10-26 23:14:37 +00:00
pc Reland: Reject the description with fewer m= sections. 2017-10-27 01:07:27 +00:00
resources Fix of integer overflow in WebRtcAecm_ProcessBlock / ApmTest.Process 2017-10-23 14:25:37 +00:00
rtc_base Adding RTC_ prefixed LOG macros. 2017-10-27 08:24:37 +00:00
rtc_tools Estimate RTP clock frequency and plot capture-send delay. 2017-10-26 08:42:54 +00:00
sdk Revert ObjC API changes for BWE allocation strategy 2017-10-27 12:33:24 +00:00
stats Added RTCMediaStreamTrackStats.jitterBufferDelay for audio 2017-10-02 10:47:00 +00:00
system_wrappers Switch metrics_default and field_trial_default to source_set. 2017-10-26 11:47:46 +00:00
test Allow injection of custom network models in place of FakeNetworkPipe. 2017-10-26 11:11:25 +00:00
tools_webrtc MB: Add Android Perf (swarming) 2017-10-23 13:02:07 +00:00
video Delete VCMSendStatisticsCallback and corresponding use of ProcessThread 2017-10-26 08:13:55 +00:00
voice_engine Let ChanneOwner use scoped_refptr and RefCountedBase. 2017-10-23 14:22:17 +00:00
.clang-format Tune ObjC clang-format configuration 2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs Create .git-blame-ignore-revs and add Java format CL to it. 2016-10-20 09:20:39 +00:00
.gitignore MB: Add support for isolating scripts + isolate low_bandwidth_audio_test.py. 2017-10-02 16:57:09 +00:00
.gn Remove remaining mentions of gflags 2017-09-25 15:34:41 +00:00
.vpython Add psutil to vpython dependencies (used on builder bots) 2017-09-04 08:04:16 +00:00
AUTHORS Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ ) 2017-09-25 13:37:12 +00:00
BUILD.gn Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." 2017-09-29 13:48:29 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +00:00
codereview.settings Make Gerrit the default for WebRTC changes 2017-09-29 01:38:07 +00:00
common_types.cc Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
common_types.h Set RTPVideoHeader picture id in PayloadRouter if forced fallback for VP8 is enabled. 2017-10-06 13:41:14 +00:00
DEPS Roll chromium_revision 318723cc57..6e83f76015 (512265:512266) 2017-10-27 21:22:12 +00:00
LICENSE Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt Update template to follow chromium copyright style 2013-04-24 01:01:28 +00:00
LICENSE_THIRD_PARTY Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
OWNERS Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Presubmit: Don't forget to warn when changing headers in subdirs of api/ 2017-10-25 11:53:16 +00:00
presubmit_test.py Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks. 2017-10-13 03:48:26 +00:00
presubmit_test_mocks.py Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks. 2017-10-13 03:48:26 +00:00
pylintrc Removing invalid-name from disabled pylint checks. 2017-10-11 08:06:49 +00:00
README.chromium Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
README.md Update README.md and codereview.settings for new source location 2017-09-13 19:54:59 +00:00
style-guide.md Style guide: Attempt to make the L2 and L3 headings more visually distinct 2017-09-09 03:52:23 +00:00
typedefs.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
WATCHLISTS Add myself to the watchlist for webrtc/api/ and webrtc/base/ 2017-05-04 13:22:46 +00:00
webrtc.gni Remove QUIC transport/data channel 2017-10-24 16:14:18 +00:00
whitespace.txt Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00

README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

More info