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Make UBSan warnings fatal and fix the existing ones

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The warnings were (all signed integer overflow):
webrtc/common_audio/signal_processing/levinson_durbin.c:46:25
12 * 268435456 cannot be represented in type 'int'
webrtc/modules/audio_processing/aecm/aecm_core.cc:930:69
522240 * 6115 cannot be represented in type 'int'
webrtc/modules/audio_processing/aecm/aecm_core_c.cc:455:36
72293096 * 50 cannot be represented in type 'int'
webrtc/modules/pacing/alr_detector.cc:70:48
1000000000 * 65 cannot be represented in type 'int'
webrtc/modules/rtp_rtcp/source/rtp_sender.cc:947:20
1929277286 + 321546521 cannot be represented in type 'int'

BUG=webrtc:8195

Review-Url: https://codereview.webrtc.org/3005003002
Cr-Commit-Position: refs/heads/master@{#19670}
This commit is contained in:
oprypin
2017-09-04 08:32:43 -07:00
committed by Commit Bot
parent 9762d4f2e5
commit ba09f79ba3
5 changed files with 16 additions and 9 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
tools_webrtc/mb/mb_config.pyl
View File

@ -461,11 +461,11 @@
},
'ubsan': {
'gn_args': 'is_ubsan=true',
'gn_args': 'is_ubsan=true is_ubsan_no_recover=true',
},
'ubsan_vptr': {
'gn_args': 'is_ubsan_vptr=true',
'gn_args': 'is_ubsan_vptr=true is_ubsan_no_recover=true',
},
'win_fastlink': {

5
tools_webrtc/ubsan/blacklist.txt
View File

@ -22,6 +22,10 @@ src:*/usr/*
# https://bugs.chromium.org/p/webrtc/issues/detail?id=5513
fun:*FilterBanksTest*CalculateResidualEnergyTester*
#############################################################################
# https://bugs.chromium.org/p/webrtc/issues/detail?id=8200
fun:*WebRtcAecm_ProcessBlock*
#############################################################################
# Ignore errors in common_audio.
# https://bugs.chromium.org/p/webrtc/issues/detail?id=5486
@ -30,4 +34,5 @@ src:*/webrtc/common_audio/signal_processing/resample_by_2_internal.c
fun:*WebRtcSpl_AddSatW32*
fun:*WebRtcSpl_SubSatW32*
fun:*WebRtcSpl_DivW32HiLow*
fun:*WebRtcSpl_LevinsonDurbin*
fun:*GmmProbability*

5
webrtc/modules/audio_processing/aecm/aecm_core.cc
View File

@ -24,6 +24,7 @@ extern "C" {
}
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/safe_conversions.h"
#include "webrtc/typedefs.h"
#ifdef AEC_DEBUG
@ -927,8 +928,8 @@ void WebRtcAecm_UpdateChannel(AecmCore* aecm,
{
// We need to shift down before multiplication
shiftChFar = 32 - zerosCh - zerosFar;
tmpU32no1 = (aecm->channelAdapt32[i] >> shiftChFar) *
far_spectrum[i];
tmpU32no1 = rtc::dchecked_cast<uint32_t>(
aecm->channelAdapt32[i] >> shiftChFar) * far_spectrum[i];
}
// Determine Q-domain of numerator
zerosNum = WebRtcSpl_NormU32(tmpU32no1);

5
webrtc/modules/pacing/alr_detector.cc
View File

@ -67,8 +67,9 @@ void AlrDetector::OnBytesSent(size_t bytes_sent, int64_t delta_time_ms) {
void AlrDetector::SetEstimatedBitrate(int bitrate_bps) {
RTC_DCHECK(bitrate_bps);
alr_budget_.set_target_rate_kbps(bitrate_bps * bandwidth_usage_percent_ /
(1000 * 100));
const auto target_rate_kbps = int64_t{bitrate_bps} *
bandwidth_usage_percent_ / (1000 * 100);
alr_budget_.set_target_rate_kbps(rtc::dchecked_cast<int>(target_rate_kbps));
}
rtc::Optional<int64_t> AlrDetector::GetApplicationLimitedRegionStartTime()

6
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -936,7 +936,7 @@ void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
return;
uint32_t ssrc;
int avg_delay_ms = 0;
int64_t avg_delay_ms = 0;
int max_delay_ms = 0;
{
rtc::CritScope lock(&send_critsect_);
@ -962,8 +962,8 @@ void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
return;
avg_delay_ms = (avg_delay_ms + num_delays / 2) / num_delays;
}
send_side_delay_observer_->SendSideDelayUpdated(avg_delay_ms, max_delay_ms,
ssrc);
send_side_delay_observer_->SendSideDelayUpdated(
rtc::dchecked_cast<int>(avg_delay_ms), max_delay_ms, ssrc);
}
void RTPSender::UpdateOnSendPacket(int packet_id,