Index: webrtc/call/rtp_transport_controller_send_interface.h |
diff --git a/webrtc/call/rtp_transport_controller_send_interface.h b/webrtc/call/rtp_transport_controller_send_interface.h |
index bd71da02e948db6aea77e5b9d75d924a9784846d..b58042197ebea8ad0184a53ece0a5d61477527d4 100644 |
--- a/webrtc/call/rtp_transport_controller_send_interface.h |
+++ b/webrtc/call/rtp_transport_controller_send_interface.h |
@@ -13,6 +13,7 @@ |
namespace webrtc { |
+class PacedSender; |
class PacketRouter; |
class RtpPacketSender; |
struct RtpKeepAliveConfig; |
@@ -46,12 +47,25 @@ class RtpTransportControllerSendInterface { |
public: |
virtual ~RtpTransportControllerSendInterface() {} |
virtual PacketRouter* packet_router() = 0; |
+ virtual PacedSender* pacer() = 0; |
// Currently returning the same pointer, but with different types. |
virtual SendSideCongestionController* send_side_cc() = 0; |
virtual TransportFeedbackObserver* transport_feedback_observer() = 0; |
virtual RtpPacketSender* packet_sender() = 0; |
virtual const RtpKeepAliveConfig& keepalive_config() const = 0; |
+ |
+ // SetAllocatedSendBitrateLimits sets bitrates limits imposed by send codec |
+ // settings. |
+ // |min_send_bitrate_bps| is the total minimum send bitrate required by all |
+ // sending streams. This is the minimum bitrate the PacedSender will use. |
+ // Note that SendSideCongestionController::OnNetworkChanged can still be |
+ // called with a lower bitrate estimate. |max_padding_bitrate_bps| is the max |
+ // bitrate the send streams request for padding. This can be higher than the |
+ // current network estimate and tells the PacedSender how much it should max |
+ // pad unless there is real packets to send. |
+ virtual void SetAllocatedSendBitrateLimits(int min_send_bitrate_bps, |
+ int max_padding_bitrate_bps) = 0; |
}; |
} // namespace webrtc |