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