Chromium Code Reviews| Index: webrtc/call/BUILD.gn | 
| diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn | 
| index 64829e9b9d43e8ff07c023523b6ca65bf3449210..5572941f30efeec1f7d8497b333ef4d24ec7f3de 100644 | 
| --- a/webrtc/call/BUILD.gn | 
| +++ b/webrtc/call/BUILD.gn | 
| @@ -8,6 +8,16 @@ | 
| import("../build/webrtc.gni") | 
| +rtc_source_set("call_interfaces") { | 
| + sources = [ | 
| + "audio_receive_stream.h", | 
| + "audio_send_stream.cc", | 
| 
 
ossu
2016/12/06 16:05:02
Since call.h and call.cc are separated between cal
 
the sun
2016/12/06 21:43:22
The audio_send_stream.cc here defines default ctor
 
 | 
| + "audio_send_stream.h", | 
| + "audio_state.h", | 
| + "call.h", | 
| + ] | 
| +} | 
| + | 
| rtc_static_library("call") { | 
| sources = [ | 
| "bitrate_allocator.cc", | 
| @@ -22,10 +32,12 @@ rtc_static_library("call") { | 
| } | 
| public_deps = [ | 
| + ":call_interfaces", | 
| "../api:call_api", | 
| ] | 
| deps = [ | 
| + ":call_interfaces", | 
| "..:webrtc_common", | 
| "../api:transport_api", | 
| "../audio", |