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In ReceiveStatistics fix a signed integer overflow undefined behavior

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Bug: b/318332290
Change-Id: I279dcaf8c9cb801482f0e29343304c854af78792
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333060
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41463}
This commit is contained in:
Danil Chapovalov 2024-01-02 11:54:28 +01:00 committed by WebRTC LUCI CQ
parent 764ac7ec0a
commit 8a74636d46
2 changed files with 21 additions and 4 deletions
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8
modules/rtp_rtcp/source/receive_statistics_impl.cc
View File

@ -18,6 +18,7 @@
#include "api/units/time_delta.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
@ -159,16 +160,15 @@ void StreamStatisticianImpl::UpdateJitter(const RtpPacketReceived& packet,
int32_t time_diff_samples =
receive_diff_rtp - (packet.Timestamp() - last_received_timestamp_);
time_diff_samples = std::abs(time_diff_samples);
ReviseFrequencyAndJitter(packet.payload_type_frequency());
// lib_jingle sometimes deliver crazy jumps in TS for the same stream.
// If this happens, don't update jitter value. Use 5 secs video frequency
// as the threshold.
if (time_diff_samples < 450000) {
if (time_diff_samples < 5 * kVideoPayloadTypeFrequency &&
time_diff_samples > -5 * kVideoPayloadTypeFrequency) {
// Note we calculate in Q4 to avoid using float.
int32_t jitter_diff_q4 = (time_diff_samples << 4) - jitter_q4_;
int32_t jitter_diff_q4 = (std::abs(time_diff_samples) << 4) - jitter_q4_;
jitter_q4_ += ((jitter_diff_q4 + 8) >> 4);
}
}

17
modules/rtp_rtcp/source/receive_statistics_unittest.cc
View File

@ -898,5 +898,22 @@ TEST(ReviseJitterTest,
EXPECT_EQ(GetJitter(*statistics), 172U);
}
TEST(ReviseJitterTest, TwoPacketsWithMaximumRtpTimestampDifference) {
SimulatedClock clock(0);
std::unique_ptr<ReceiveStatistics> statistics =
ReceiveStatistics::Create(&clock);
RtpPacketReceived packet1 = MakeRtpPacket(/*payload_type_frequency=*/90'000,
/*timestamp=*/0x01234567);
RtpPacketReceived packet2 =
MakeNextRtpPacket(packet1,
/*payload_type_frequency=*/90'000,
/*timestamp=*/0x81234567);
statistics->OnRtpPacket(packet1);
statistics->OnRtpPacket(packet2);
// Expect large jump in RTP timestamp is ignored for jitter calculation.
EXPECT_EQ(GetJitter(*statistics), 0U);
}
} // namespace
} // namespace webrtc