7bb8f02274
This is accomplished by breaking out RTX and FEC/RED functionality from the RTP module and keeping track of the base payload type, that is the payload type received when not receiving RTX. Enables retransmissions over RTX by default in the loopback test. BUG=1811 TESTS=voe/vie_auto_test --automated and trybots. R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2154004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4692 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);
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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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