Delete or update left-over ASSERT use and comments.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 478 matching lines @@
       }
     }
     return selected_packet_size_ms;
   }
 
   // Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC
   // which says that G722 should be advertised as 8 kHz although it is a 16 kHz
   // codec.
   static void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
     if (IsCodec(*voe_codec, kG722CodecName)) {
-      // If the ASSERT triggers, the codec definition in WebRTC VoiceEngine
+      // If the DCHECK triggers, the codec definition in WebRTC VoiceEngine
       // has changed, and this special case is no longer needed.
       RTC_DCHECK(voe_codec->plfreq != new_plfreq);
       voe_codec->plfreq = new_plfreq;
     }
   }
 };
 
 const WebRtcVoiceCodecs::CodecPref WebRtcVoiceCodecs::kCodecPrefs[14] = {
     {kOpusCodecName, 48000, 2, 111, true, {10, 20, 40, 60}, kOpusMaxBitrateBps},
     {kIsacCodecName, 16000, 1, 103, true, {30, 60}, kIsacMaxBitrateBps},
@@ skipping 2203 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE