| Index: webrtc/call/call.cc
|
| diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
|
| index 0e64269d4e4c70391a00be65da2440778bb14197..241a6be6607f0381b25abbca867a236fba9f2d90 100644
|
| --- a/webrtc/call/call.cc
|
| +++ b/webrtc/call/call.cc
|
| @@ -1277,13 +1277,9 @@ void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
|
| // enabled.
|
| return;
|
| }
|
| - // For audio, we only support send side BWE.
|
| - if (media_type == MediaType::VIDEO ||
|
| - (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
|
| - receive_side_cc_.OnReceivedPacket(
|
| - packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
|
| - header);
|
| - }
|
| + receive_side_cc_.OnReceivedPacket(
|
| + packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
|
| + header);
|
| }
|
|
|
| } // namespace internal
|
|
|