Removed an obsolete DCHECK in AudioEncoderOpus.

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

 /*
  *  Copyright (c) 2014 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 711 matching lines @@
   RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, GetBitrateBps(config_)));
   const auto new_complexity = GetNewComplexity(config_);
   if (new_complexity && complexity_ != *new_complexity) {
     complexity_ = *new_complexity;
     RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
   }
 }
 
 void AudioEncoderOpus::ApplyAudioNetworkAdaptor() {
   auto config = audio_network_adaptor_->GetEncoderRuntimeConfig();
-  RTC_DCHECK(!config.frame_length_ms || *config.frame_length_ms == 20 ||
-             *config.frame_length_ms == 60);
 
   if (config.bitrate_bps)
     SetTargetBitrate(*config.bitrate_bps);
   if (config.frame_length_ms)
     SetFrameLength(*config.frame_length_ms);
   if (config.enable_fec)
     SetFec(*config.enable_fec);
   if (config.uplink_packet_loss_fraction)
     SetProjectedPacketLossRate(*config.uplink_packet_loss_fraction);
   if (config.enable_dtx)
@@ skipping 24 matching lines @@
             config_.uplink_bandwidth_update_interval_ms) {
       rtc::Optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
       if (smoothed_bitrate)
         audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
       bitrate_smoother_last_update_time_ = rtc::Optional<int64_t>(now_ms);
     }
   }
 }
 
 }  // namespace webrtc