Index: webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h |
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h |
index 14ddbca0a781e2bfd257843291d185540739e95c..0ad4a1e5f3b510380ed53da90666a40821fdf8c7 100644 |
--- a/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h |
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h |
@@ -26,6 +26,8 @@ class AudioNetworkAdaptor { |
~EncoderRuntimeConfig(); |
rtc::Optional<int> bitrate_bps; |
rtc::Optional<int> frame_length_ms; |
+ // Note: This is what we tell the encoder. It doesn't have to reflect |
+ // the actual NetworkMetrics; it's subject to our decision. |
rtc::Optional<float> uplink_packet_loss_fraction; |
rtc::Optional<bool> enable_fec; |
rtc::Optional<bool> enable_dtx; |
@@ -43,6 +45,9 @@ class AudioNetworkAdaptor { |
virtual void SetUplinkPacketLossFraction( |
float uplink_packet_loss_fraction) = 0; |
+ virtual void SetUplinkRecoverablePacketLossFraction( |
+ float uplink_recoverable_packet_loss_fraction) = 0; |
+ |
virtual void SetRtt(int rtt_ms) = 0; |
virtual void SetTargetAudioBitrate(int target_audio_bitrate_bps) = 0; |