Chromium Code Reviews| Index: webrtc/call/call.cc |
| diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc |
| index bb13d558ab2afbd149f6904c4aad907f4b9fdeac..7fd131e4c823d6effd390ed678d9aec956e60c65 100644 |
| --- a/webrtc/call/call.cc |
| +++ b/webrtc/call/call.cc |
| @@ -1223,6 +1223,9 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type, |
| } |
| } |
| if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) { |
| + received_bytes_per_second_counter_.Add(static_cast<int>(length)); |
|
stefan-webrtc
2017/02/21 12:06:37
Should this be a size_t counter?
brandtr
2017/02/22 08:27:59
Maybe? The container only supports int however: ht
|
| + // TODO(brandtr): Update here when FlexFEC supports protecting audio. |
| + received_video_bytes_per_second_counter_.Add(static_cast<int>(length)); |
| auto it = flexfec_receive_ssrcs_protection_.find(ssrc); |
| if (it != flexfec_receive_ssrcs_protection_.end()) { |
| it->second->OnRtpPacket(*parsed_packet); |