822fbd8b68
Together with Stefan's http://review.webrtc.org/1960004/. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2048004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
119 lines
3.6 KiB
C++
119 lines
3.6 KiB
C++
/*
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* Copyright (c) 2012 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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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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namespace webrtc {
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// This class sends all its packet straight to the provided RtpRtcp module.
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// with optional packet loss.
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class LoopBackTransport : public webrtc::Transport {
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public:
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LoopBackTransport()
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: _count(0),
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_packetLoss(0),
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rtp_payload_registry_(NULL),
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rtp_receiver_(NULL),
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_rtpRtcpModule(NULL) {
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}
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void SetSendModule(RtpRtcp* rtpRtcpModule,
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RTPPayloadRegistry* payload_registry,
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RtpReceiver* receiver,
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ReceiveStatistics* receive_statistics) {
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_rtpRtcpModule = rtpRtcpModule;
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rtp_payload_registry_ = payload_registry;
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rtp_receiver_ = receiver;
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receive_statistics_ = receive_statistics;
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}
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void DropEveryNthPacket(int n) {
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_packetLoss = n;
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}
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virtual int SendPacket(int channel, const void *data, int len) {
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_count++;
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if (_packetLoss > 0) {
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if ((_count % _packetLoss) == 0) {
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return len;
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}
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}
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RTPHeader header;
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scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
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if (!parser->Parse(static_cast<const uint8_t*>(data), len, &header)) {
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return -1;
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}
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PayloadUnion payload_specific;
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if (!rtp_payload_registry_->GetPayloadSpecifics(
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header.payloadType, &payload_specific)) {
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return -1;
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}
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receive_statistics_->IncomingPacket(header, len, false, true);
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if (!rtp_receiver_->IncomingRtpPacket(&header,
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static_cast<const uint8_t*>(data),
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len, payload_specific, true)) {
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return -1;
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}
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return len;
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}
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virtual int SendRTCPPacket(int channel, const void *data, int len) {
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if (_rtpRtcpModule->IncomingRtcpPacket((const uint8_t*)data, len) < 0) {
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return -1;
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}
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return len;
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}
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private:
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int _count;
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int _packetLoss;
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ReceiveStatistics* receive_statistics_;
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RTPPayloadRegistry* rtp_payload_registry_;
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RtpReceiver* rtp_receiver_;
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RtpRtcp* _rtpRtcpModule;
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};
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class TestRtpReceiver : public NullRtpData {
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public:
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virtual int32_t OnReceivedPayloadData(
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const uint8_t* payloadData,
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const uint16_t payloadSize,
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const webrtc::WebRtcRTPHeader* rtpHeader) {
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EXPECT_LE(payloadSize, sizeof(_payloadData));
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memcpy(_payloadData, payloadData, payloadSize);
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memcpy(&_rtpHeader, rtpHeader, sizeof(_rtpHeader));
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_payloadSize = payloadSize;
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return 0;
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}
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const uint8_t* payload_data() const {
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return _payloadData;
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}
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uint16_t payload_size() const {
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return _payloadSize;
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}
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webrtc::WebRtcRTPHeader rtp_header() const {
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return _rtpHeader;
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}
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private:
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uint8_t _payloadData[1500];
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uint16_t _payloadSize;
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webrtc::WebRtcRTPHeader _rtpHeader;
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};
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} // namespace webrtc
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