| Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
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| diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
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| index af66cd3634ee1cdabcfab57f18c80c953a0a9391..c6bac621c31e63b0fd0f4b2f29f23a735c48e983 100644
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| --- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
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| +++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
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| @@ -491,23 +491,21 @@ void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
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|  
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|  void AudioEncoderOpus::ApplyAudioNetworkAdaptor() {
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|    auto config = audio_network_adaptor_->GetEncoderRuntimeConfig();
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| -  // |audio_network_adaptor_| is supposed to be configured to output all
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| -  // following parameters.
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| -  RTC_DCHECK(config.bitrate_bps);
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| -  RTC_DCHECK(config.frame_length_ms);
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| -  RTC_DCHECK(config.uplink_packet_loss_fraction);
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| -  RTC_DCHECK(config.enable_fec);
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| -  RTC_DCHECK(config.enable_dtx);
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| -  RTC_DCHECK(config.num_channels);
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| -
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| -  RTC_DCHECK(*config.frame_length_ms == 20 || *config.frame_length_ms == 60);
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| -
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| -  SetTargetBitrate(*config.bitrate_bps);
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| -  SetFrameLength(*config.frame_length_ms);
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| -  SetFec(*config.enable_fec);
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| -  SetProjectedPacketLossRate(*config.uplink_packet_loss_fraction);
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| -  SetDtx(*config.enable_dtx);
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| -  SetNumChannelsToEncode(*config.num_channels);
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| +  RTC_DCHECK(!config.frame_length_ms || *config.frame_length_ms == 20 ||
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| +             *config.frame_length_ms == 60);
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| +
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| +  if (config.bitrate_bps)
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| +    SetTargetBitrate(*config.bitrate_bps);
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| +  if (config.frame_length_ms)
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| +    SetFrameLength(*config.frame_length_ms);
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| +  if (config.enable_fec)
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| +    SetFec(*config.enable_fec);
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| +  if (config.uplink_packet_loss_fraction)
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| +    SetProjectedPacketLossRate(*config.uplink_packet_loss_fraction);
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| +  if (config.enable_dtx)
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| +    SetDtx(*config.enable_dtx);
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| +  if (config.num_channels)
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| +    SetNumChannelsToEncode(*config.num_channels);
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|  }
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|  
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|  std::unique_ptr<AudioNetworkAdaptor>
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| 
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