|  | /* | 
|  | *  Copyright 2017 The WebRTC project authors. All Rights Reserved. | 
|  | * | 
|  | *  Use of this source code is governed by a BSD-style license | 
|  | *  that can be found in the LICENSE file in the root of the source | 
|  | *  tree. An additional intellectual property rights grant can be found | 
|  | *  in the file PATENTS.  All contributing project authors may | 
|  | *  be found in the AUTHORS file in the root of the source tree. | 
|  | */ | 
|  |  | 
|  | #ifndef PC_RTPTRANSPORTTESTUTIL_H_ | 
|  | #define PC_RTPTRANSPORTTESTUTIL_H_ | 
|  |  | 
|  | #include "pc/rtptransportinternal.h" | 
|  | #include "rtc_base/sigslot.h" | 
|  |  | 
|  | namespace webrtc { | 
|  |  | 
|  | class SignalPacketReceivedCounter : public sigslot::has_slots<> { | 
|  | public: | 
|  | explicit SignalPacketReceivedCounter(RtpTransportInternal* transport) { | 
|  | transport->SignalPacketReceived.connect( | 
|  | this, &SignalPacketReceivedCounter::OnPacketReceived); | 
|  | } | 
|  | int rtcp_count() const { return rtcp_count_; } | 
|  | int rtp_count() const { return rtp_count_; } | 
|  |  | 
|  | private: | 
|  | void OnPacketReceived(bool rtcp, | 
|  | rtc::CopyOnWriteBuffer*, | 
|  | const rtc::PacketTime&) { | 
|  | if (rtcp) { | 
|  | ++rtcp_count_; | 
|  | } else { | 
|  | ++rtp_count_; | 
|  | } | 
|  | } | 
|  | int rtcp_count_ = 0; | 
|  | int rtp_count_ = 0; | 
|  | }; | 
|  |  | 
|  | }  // namespace webrtc | 
|  |  | 
|  | #endif  // PC_RTPTRANSPORTTESTUTIL_H_ |