| Index: webrtc/call/call.h
|
| diff --git a/webrtc/call/call.h b/webrtc/call/call.h
|
| index 86142f0feaaf065c3215a9e3086bfdf63b08911e..569f6f3c9a0cc13e70e13636d73c964011c56282 100644
|
| --- a/webrtc/call/call.h
|
| +++ b/webrtc/call/call.h
|
| @@ -22,6 +22,7 @@
|
| #include "webrtc/call/flexfec_receive_stream.h"
|
| #include "webrtc/call/rtp_transport_controller_send_interface.h"
|
| #include "webrtc/common_types.h"
|
| +#include "webrtc/rtc_base/bitrateallocationstrategy.h"
|
| #include "webrtc/rtc_base/networkroute.h"
|
| #include "webrtc/rtc_base/platform_file.h"
|
| #include "webrtc/rtc_base/socket.h"
|
| @@ -183,6 +184,10 @@ class Call {
|
| virtual void SetBitrateConfigMask(
|
| const Config::BitrateConfigMask& bitrate_mask) = 0;
|
|
|
| + virtual void SetBitrateAllocationStrategy(
|
| + rtc::scoped_refptr<rtc::BitrateAllocationStrategy>
|
| + bitrate_allocation_strategy) = 0;
|
| +
|
| // TODO(skvlad): When the unbundled case with multiple streams for the same
|
| // media type going over different networks is supported, track the state
|
| // for each stream separately. Right now it's global per media type.
|
|
|