Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
index 3cac4195b46c132d58e1c968c22283fe0ae826b5..1bdb804e8226366578439ff881c30d9126c06c27 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
@@ -17,8 +17,9 @@ |
#include "webrtc/base/checks.h" |
#include "webrtc/base/logging.h" |
#include "webrtc/base/rate_limiter.h" |
-#include "webrtc/base/trace_event.h" |
+#include "webrtc/base/safe_minmax.h" |
#include "webrtc/base/timeutils.h" |
+#include "webrtc/base/trace_event.h" |
#include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h" |
#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" |
@@ -467,16 +468,16 @@ size_t RTPSender::SendPadData(size_t bytes, |
if (audio_configured_) { |
// Allow smaller padding packets for audio. |
- padding_bytes_in_packet = |
- std::min(std::max(bytes, kMinAudioPaddingLength), max_payload_size); |
- if (padding_bytes_in_packet > kMaxPaddingLength) |
- padding_bytes_in_packet = kMaxPaddingLength; |
+ padding_bytes_in_packet = rtc::SafeClamp<size_t>( |
+ bytes, kMinAudioPaddingLength, |
+ rtc::SafeMin(max_payload_size, kMaxPaddingLength)); |
} else { |
// Always send full padding packets. This is accounted for by the |
// RtpPacketSender, which will make sure we don't send too much padding even |
// if a single packet is larger than requested. |
// We do this to avoid frequently sending small packets on higher bitrates. |
- padding_bytes_in_packet = std::min(max_payload_size, kMaxPaddingLength); |
+ padding_bytes_in_packet = |
+ rtc::SafeMin<size_t>(max_payload_size, kMaxPaddingLength); |
} |
size_t bytes_sent = 0; |
while (bytes_sent < bytes) { |