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", |