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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
44 lines
1.2 KiB
C++
44 lines
1.2 KiB
C++
/*
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* Copyright 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef PC_RTPTRANSPORTTESTUTIL_H_
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#define PC_RTPTRANSPORTTESTUTIL_H_
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#include "pc/rtptransportinternal.h"
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#include "rtc_base/sigslot.h"
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namespace webrtc {
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class SignalPacketReceivedCounter : public sigslot::has_slots<> {
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public:
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explicit SignalPacketReceivedCounter(RtpTransportInternal* transport) {
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transport->SignalPacketReceived.connect(
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this, &SignalPacketReceivedCounter::OnPacketReceived);
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}
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int rtcp_count() const { return rtcp_count_; }
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int rtp_count() const { return rtp_count_; }
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private:
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void OnPacketReceived(bool rtcp,
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rtc::CopyOnWriteBuffer*,
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const rtc::PacketTime&) {
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if (rtcp) {
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++rtcp_count_;
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} else {
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++rtp_count_;
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}
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}
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int rtcp_count_ = 0;
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int rtp_count_ = 0;
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};
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} // namespace webrtc
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#endif // PC_RTPTRANSPORTTESTUTIL_H_
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