| /* |
| * 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/base/scoped_ptr.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" |
| |
| 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), |
| packet_loss_(0), |
| rtp_payload_registry_(NULL), |
| rtp_receiver_(NULL), |
| rtp_rtcp_module_(NULL) {} |
| void SetSendModule(RtpRtcp* rtp_rtcp_module, |
| RTPPayloadRegistry* payload_registry, |
| RtpReceiver* receiver, |
| ReceiveStatistics* receive_statistics); |
| void DropEveryNthPacket(int n); |
| int SendPacket(int channel, const void* data, size_t len) override; |
| int SendRTCPPacket(int channel, const void* data, size_t len) override; |
| |
| private: |
| int count_; |
| int packet_loss_; |
| ReceiveStatistics* receive_statistics_; |
| RTPPayloadRegistry* rtp_payload_registry_; |
| RtpReceiver* rtp_receiver_; |
| RtpRtcp* rtp_rtcp_module_; |
| }; |
| |
| class TestRtpReceiver : public NullRtpData { |
| public: |
| int32_t OnReceivedPayloadData( |
| const uint8_t* payload_data, |
| const size_t payload_size, |
| const webrtc::WebRtcRTPHeader* rtp_header) override; |
| |
| const uint8_t* payload_data() const { return payload_data_; } |
| size_t payload_size() const { return payload_size_; } |
| webrtc::WebRtcRTPHeader rtp_header() const { return rtp_header_; } |
| |
| private: |
| uint8_t payload_data_[1500]; |
| size_t payload_size_; |
| webrtc::WebRtcRTPHeader rtp_header_; |
| }; |
| |
| } // namespace webrtc |