/* * Copyright (c) 2012 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. */ #include "common_types.h" #include "rtp_rtcp.h" #include "rtp_rtcp_defines.h" namespace webrtc { class FakeRtpRtcpClock : public RtpRtcpClock { public: FakeRtpRtcpClock() { time_in_ms_ = 123456; } // Return a timestamp in milliseconds relative to some arbitrary // source; the source is fixed for this clock. virtual WebRtc_Word64 GetTimeInMS() { return time_in_ms_; } // Retrieve an NTP absolute timestamp. virtual void CurrentNTP(WebRtc_UWord32& secs, WebRtc_UWord32& frac) { secs = time_in_ms_ / 1000; frac = (time_in_ms_ % 1000) * 4294967; } void IncrementTime(WebRtc_UWord32 time_increment_ms) { time_in_ms_ += time_increment_ms; } private: WebRtc_Word64 time_in_ms_; }; // This class sends all its packet straight to the provided RtpRtcp module. // with optional packet loss. class LoopBackTransport : public webrtc::Transport { public: LoopBackTransport() : _count(0), _packetLoss(0), _rtpRtcpModule(NULL) { } void SetSendModule(RtpRtcp* rtpRtcpModule) { _rtpRtcpModule = rtpRtcpModule; } void DropEveryNthPacket(int n) { _packetLoss = n; } virtual int SendPacket(int channel, const void *data, int len) { _count++; if (_packetLoss > 0) { if ((_count % _packetLoss) == 0) { return len; } } if (_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data, len) == 0) { return len; } return -1; } virtual int SendRTCPPacket(int channel, const void *data, int len) { if (_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data, len) == 0) { return len; } return -1; } private: int _count; int _packetLoss; RtpRtcp* _rtpRtcpModule; }; class RtpReceiver : public RtpData { public: virtual WebRtc_Word32 OnReceivedPayloadData( const WebRtc_UWord8* payloadData, const WebRtc_UWord16 payloadSize, const webrtc::WebRtcRTPHeader* rtpHeader) { return 0; } }; } // namespace webrtc