92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
73 lines
2.4 KiB
C++
73 lines
2.4 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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#ifndef MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
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#define MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
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#include "api/call/transport.h"
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#include "common_types.h"
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#include "modules/rtp_rtcp/include/receive_statistics.h"
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#include "modules/rtp_rtcp/include/rtp_header_parser.h"
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#include "modules/rtp_rtcp/include/rtp_payload_registry.h"
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#include "modules/rtp_rtcp/include/rtp_receiver.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "test/gtest.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 Transport {
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public:
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LoopBackTransport()
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: count_(0),
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packet_loss_(0),
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rtp_payload_registry_(NULL),
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rtp_receiver_(NULL),
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rtp_rtcp_module_(NULL) {}
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void SetSendModule(RtpRtcp* rtp_rtcp_module,
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RTPPayloadRegistry* payload_registry,
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RtpReceiver* receiver,
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ReceiveStatistics* receive_statistics);
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void DropEveryNthPacket(int n);
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bool SendRtp(const uint8_t* data,
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size_t len,
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const PacketOptions& options) override;
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bool SendRtcp(const uint8_t* data, size_t len) override;
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private:
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int count_;
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int packet_loss_;
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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* rtp_rtcp_module_;
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};
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class TestRtpReceiver : public RtpData {
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public:
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int32_t OnReceivedPayloadData(
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const uint8_t* payload_data,
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size_t payload_size,
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const webrtc::WebRtcRTPHeader* rtp_header) override;
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const uint8_t* payload_data() const { return payload_data_; }
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size_t payload_size() const { return payload_size_; }
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webrtc::WebRtcRTPHeader rtp_header() const { return rtp_header_; }
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private:
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uint8_t payload_data_[1500];
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size_t payload_size_;
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webrtc::WebRtcRTPHeader rtp_header_;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
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