/*
* 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, int len) {
_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),
static_cast<size_t>(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, int len) {
if (_rtpRtcpModule->IncomingRtcpPacket((const uint8_t*)data, len) < 0) {
return -1;
}
return 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 uint16_t payloadSize,
const webrtc::WebRtcRTPHeader* rtpHeader) {
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;
}
uint16_t payload_size() const {
return _payloadSize;
}
webrtc::WebRtcRTPHeader rtp_header() const {
return _rtpHeader;
}
private:
uint8_t _payloadData[1500];
uint16_t _payloadSize;
webrtc::WebRtcRTPHeader _rtpHeader;
};
} // namespace webrtc
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