| /* |
| * 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 "testing/gtest/include/gtest/gtest.h" |
| #include "webrtc/common_types.h" |
| #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" |
| #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| |
| namespace webrtc { |
| |
| // 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), |
| rtp_payload_registry_(NULL), |
| rtp_receiver_(NULL), |
| _rtpRtcpModule(NULL) { |
| } |
| void SetSendModule(RtpRtcp* rtpRtcpModule, |
| RTPPayloadRegistry* payload_registry, |
| RtpReceiver* receiver, |
| ReceiveStatistics* receive_statistics) { |
| _rtpRtcpModule = rtpRtcpModule; |
| rtp_payload_registry_ = payload_registry; |
| rtp_receiver_ = receiver; |
| receive_statistics_ = receive_statistics; |
| } |
| void DropEveryNthPacket(int n) { |
| _packetLoss = n; |
| } |
| virtual int SendPacket(int channel, const void *data, size_t len) OVERRIDE { |
| _count++; |
| if (_packetLoss > 0) { |
| if ((_count % _packetLoss) == 0) { |
| return len; |
| } |
| } |
| RTPHeader header; |
| scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); |
| if (!parser->Parse(static_cast<const uint8_t*>(data), len, &header)) { |
| return -1; |
| } |
| PayloadUnion payload_specific; |
| if (!rtp_payload_registry_->GetPayloadSpecifics( |
| header.payloadType, &payload_specific)) { |
| return -1; |
| } |
| receive_statistics_->IncomingPacket(header, len, false); |
| if (!rtp_receiver_->IncomingRtpPacket(header, |
| static_cast<const uint8_t*>(data), |
| len, payload_specific, true)) { |
| return -1; |
| } |
| return len; |
| } |
| virtual int SendRTCPPacket(int channel, |
| const void *data, |
| size_t len) OVERRIDE { |
| if (_rtpRtcpModule->IncomingRtcpPacket((const uint8_t*)data, len) < 0) { |
| return -1; |
| } |
| return static_cast<int>(len); |
| } |
| private: |
| int _count; |
| int _packetLoss; |
| ReceiveStatistics* receive_statistics_; |
| RTPPayloadRegistry* rtp_payload_registry_; |
| RtpReceiver* rtp_receiver_; |
| RtpRtcp* _rtpRtcpModule; |
| }; |
| |
| class TestRtpReceiver : public NullRtpData { |
| public: |
| |
| virtual int32_t OnReceivedPayloadData( |
| const uint8_t* payloadData, |
| const size_t payloadSize, |
| const webrtc::WebRtcRTPHeader* rtpHeader) OVERRIDE { |
| EXPECT_LE(payloadSize, sizeof(_payloadData)); |
| memcpy(_payloadData, payloadData, payloadSize); |
| memcpy(&_rtpHeader, rtpHeader, sizeof(_rtpHeader)); |
| _payloadSize = payloadSize; |
| return 0; |
| } |
| |
| const uint8_t* payload_data() const { |
| return _payloadData; |
| } |
| |
| size_t payload_size() const { |
| return _payloadSize; |
| } |
| |
| webrtc::WebRtcRTPHeader rtp_header() const { |
| return _rtpHeader; |
| } |
| |
| private: |
| uint8_t _payloadData[1500]; |
| size_t _payloadSize; |
| webrtc::WebRtcRTPHeader _rtpHeader; |
| }; |
| |
| } // namespace webrtc |