Add SafeClamp(), which accepts args of different types

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 
 #include <algorithm>
 #include <utility>
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/rate_limiter.h"
+#include "webrtc/base/safe_minmax.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/base/timeutils.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"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "webrtc/modules/rtp_rtcp/source/time_util.h"
@@ skipping 428 matching lines @@
   return bytes_to_send - bytes_left;
 }
 
 size_t RTPSender::SendPadData(size_t bytes,
                               const PacedPacketInfo& pacing_info) {
   size_t padding_bytes_in_packet;
   size_t max_payload_size = max_packet_size_ - RtpHeaderLength();
 
   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) {
     int64_t now_ms = clock_->TimeInMilliseconds();
     uint32_t ssrc;
     uint32_t timestamp;
     int64_t capture_time_ms;
     uint16_t sequence_number;
     int payload_type;
     bool over_rtx;
@@ skipping 777 matching lines @@
     if (rtp_overhead_bytes_per_packet_ == packet.headers_size()) {
       return;
     }
     rtp_overhead_bytes_per_packet_ = packet.headers_size();
     overhead_bytes_per_packet = rtp_overhead_bytes_per_packet_;
   }
   overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
 }
 
 }  // namespace webrtc