| /* |
| * 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 WEBRTC_PC_RTPTRANSPORTTESTUTIL_H_ |
| #define WEBRTC_PC_RTPTRANSPORTTESTUTIL_H_ |
| |
| #include "webrtc/pc/rtptransportinternal.h" |
| #include "webrtc/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 // WEBRTC_PC_RTPTRANSPORTTESTUTIL_H_ |