398c3fd6c2
SrtpTransport currently just delegates everything to RtpTransport. Also makes BaseChannel::rtp_transport_ an RtpTransportInternal and constructs an SrtpTransport if srtp is required. BUG=webrtc:7013 Review-Url: https://codereview.webrtc.org/2981013002 Cr-Commit-Position: refs/heads/master@{#19095}
45 lines
1.3 KiB
C++
45 lines
1.3 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 WEBRTC_PC_RTPTRANSPORTTESTUTIL_H_
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#define WEBRTC_PC_RTPTRANSPORTTESTUTIL_H_
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#include "webrtc/pc/rtptransportinternal.h"
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#include "webrtc/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 // WEBRTC_PC_RTPTRANSPORTTESTUTIL_H_
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