| 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
|
|
|