Introduce dchecked_cast, and start using it

webrtc/modules/audio_coding/codecs/audio_format_conversion.cc

 /*
  *  Copyright (c) 2016 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.
  */
 
@@ skipping 16 matching lines @@
                         int sample_rate,
                         int num_channels) {
   // Create a CodecInst with some fields set. The remaining fields are zeroed,
   // but we tell MSan to consider them uninitialized.
   CodecInst ci = {0};
   rtc::MsanMarkUninitialized(rtc::MakeArrayView(&ci, 1));
   ci.pltype = payload_type;
   strncpy(ci.plname, name, sizeof(ci.plname));
   ci.plname[sizeof(ci.plname) - 1] = '\0';
   ci.plfreq = sample_rate;
-  ci.channels = rtc::checked_cast<size_t>(num_channels);
+  ci.channels = rtc::dchecked_cast<size_t>(num_channels);
   return ci;
 }
 
 }  // namespace
 
 SdpAudioFormat CodecInstToSdp(const CodecInst& ci) {
   if (STR_CASE_CMP(ci.plname, "g722") == 0) {
     RTC_CHECK_EQ(16000, ci.plfreq);
     RTC_CHECK(ci.channels == 1 || ci.channels == 2);
     return {"g722", 8000, static_cast<int>(ci.channels)};
@@ skipping 31 matching lines @@
       return 1;  // Default to mono.
     }();
     return MakeCodecInst(payload_type, "opus", 48000, num_channels);
   } else {
     return MakeCodecInst(payload_type, audio_format.name.c_str(),
                          audio_format.clockrate_hz, audio_format.num_channels);
   }
 }
 
 }  // namespace webrtc