2f44673d66
BUG=314 Review URL: https://webrtc-codereview.appspot.com/1279007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3777 4adac7df-926f-26a2-2b94-8c16560cd09d
284 lines
11 KiB
C++
284 lines
11 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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/*
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* This file includes unit tests for the RTPSender.
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*/
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#include <gtest/gtest.h>
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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namespace {
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const int kId = 1;
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const int kTypeLength = TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES;
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const int kPayload = 100;
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const uint32_t kTimestamp = 10;
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const uint16_t kSeqNum = 33;
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const int kTimeOffset = 22222;
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const int kMaxPacketLength = 1500;
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} // namespace
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class LoopbackTransportTest : public webrtc::Transport {
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public:
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LoopbackTransportTest()
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: packets_sent_(0),
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last_sent_packet_len_(0) {
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}
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virtual int SendPacket(int channel, const void *data, int len) {
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packets_sent_++;
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memcpy(last_sent_packet_, data, len);
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last_sent_packet_len_ = len;
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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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return -1;
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}
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int packets_sent_;
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int last_sent_packet_len_;
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uint8_t last_sent_packet_[kMaxPacketLength];
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};
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class RtpSenderTest : public ::testing::Test {
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protected:
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RtpSenderTest()
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: fake_clock_(123456),
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rtp_sender_(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
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NULL)),
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kMarkerBit(true),
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kType(kRtpExtensionTransmissionTimeOffset) {
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rtp_sender_->SetSequenceNumber(kSeqNum);
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}
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SimulatedClock fake_clock_;
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scoped_ptr<RTPSender> rtp_sender_;
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LoopbackTransportTest transport_;
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const bool kMarkerBit;
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RTPExtensionType kType;
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uint8_t packet_[kMaxPacketLength];
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void VerifyRTPHeaderCommon(const WebRtcRTPHeader& rtp_header) {
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EXPECT_EQ(kMarkerBit, rtp_header.header.markerBit);
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EXPECT_EQ(kPayload, rtp_header.header.payloadType);
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EXPECT_EQ(kSeqNum, rtp_header.header.sequenceNumber);
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EXPECT_EQ(kTimestamp, rtp_header.header.timestamp);
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EXPECT_EQ(rtp_sender_->SSRC(), rtp_header.header.ssrc);
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EXPECT_EQ(0, rtp_header.header.numCSRCs);
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EXPECT_EQ(0, rtp_header.header.paddingLength);
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}
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};
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TEST_F(RtpSenderTest, RegisterRtpHeaderExtension) {
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EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength());
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EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kType, kId));
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EXPECT_EQ(RTP_ONE_BYTE_HEADER_LENGTH_IN_BYTES + kTypeLength,
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rtp_sender_->RtpHeaderExtensionTotalLength());
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EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(kType));
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EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength());
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}
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TEST_F(RtpSenderTest, BuildRTPPacket) {
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int32_t length = rtp_sender_->BuildRTPheader(packet_,
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kPayload,
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kMarkerBit,
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kTimestamp);
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EXPECT_EQ(12, length);
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// Verify
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webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length);
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webrtc::WebRtcRTPHeader rtp_header;
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RtpHeaderExtensionMap map;
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map.Register(kType, kId);
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const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
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ASSERT_TRUE(valid_rtp_header);
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ASSERT_FALSE(rtp_parser.RTCP());
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VerifyRTPHeaderCommon(rtp_header);
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EXPECT_EQ(length, rtp_header.header.headerLength);
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EXPECT_EQ(0, rtp_header.extension.transmissionTimeOffset);
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}
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TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) {
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EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset));
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EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kType, kId));
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int32_t length = rtp_sender_->BuildRTPheader(packet_,
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kPayload,
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kMarkerBit,
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kTimestamp);
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EXPECT_EQ(12 + rtp_sender_->RtpHeaderExtensionTotalLength(), length);
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// Verify
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webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length);
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webrtc::WebRtcRTPHeader rtp_header;
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RtpHeaderExtensionMap map;
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map.Register(kType, kId);
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const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
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ASSERT_TRUE(valid_rtp_header);
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ASSERT_FALSE(rtp_parser.RTCP());
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VerifyRTPHeaderCommon(rtp_header);
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EXPECT_EQ(length, rtp_header.header.headerLength);
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EXPECT_EQ(kTimeOffset, rtp_header.extension.transmissionTimeOffset);
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// Parse without map extension
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webrtc::WebRtcRTPHeader rtp_header2;
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const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, NULL);
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ASSERT_TRUE(valid_rtp_header2);
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VerifyRTPHeaderCommon(rtp_header2);
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EXPECT_EQ(length, rtp_header2.header.headerLength);
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EXPECT_EQ(0, rtp_header2.extension.transmissionTimeOffset);
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}
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TEST_F(RtpSenderTest, BuildRTPPacketWithNegativeTransmissionOffsetExtension) {
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const int kNegTimeOffset = -500;
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EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kNegTimeOffset));
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EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kType, kId));
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int32_t length = rtp_sender_->BuildRTPheader(packet_,
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kPayload,
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kMarkerBit,
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kTimestamp);
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EXPECT_EQ(12 + rtp_sender_->RtpHeaderExtensionTotalLength(), length);
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// Verify
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webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(packet_, length);
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webrtc::WebRtcRTPHeader rtp_header;
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RtpHeaderExtensionMap map;
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map.Register(kType, kId);
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const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
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ASSERT_TRUE(valid_rtp_header);
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ASSERT_FALSE(rtp_parser.RTCP());
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VerifyRTPHeaderCommon(rtp_header);
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EXPECT_EQ(length, rtp_header.header.headerLength);
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EXPECT_EQ(kNegTimeOffset, rtp_header.extension.transmissionTimeOffset);
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}
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TEST_F(RtpSenderTest, NoTrafficSmoothing) {
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int32_t rtp_length = rtp_sender_->BuildRTPheader(packet_,
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kPayload,
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kMarkerBit,
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kTimestamp);
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// Packet should be sent immediately.
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EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_,
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0,
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rtp_length,
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kTimestamp / 90,
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kAllowRetransmission));
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EXPECT_EQ(1, transport_.packets_sent_);
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EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_);
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}
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TEST_F(RtpSenderTest, DISABLED_TrafficSmoothing) {
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// TODO(pwestin) we need to send in a pacer object.
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rtp_sender_->SetStorePacketsStatus(true, 10);
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EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kType, kId));
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rtp_sender_->SetTargetSendBitrate(300000);
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int32_t rtp_length = rtp_sender_->BuildRTPheader(packet_,
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kPayload,
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kMarkerBit,
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kTimestamp);
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// Packet should be stored in a send bucket.
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EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_,
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0,
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rtp_length,
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fake_clock_.TimeInMilliseconds(),
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kAllowRetransmission));
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EXPECT_EQ(0, transport_.packets_sent_);
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const int kStoredTimeInMs = 100;
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fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
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// Process send bucket. Packet should now be sent.
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EXPECT_EQ(1, transport_.packets_sent_);
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EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_);
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// Parse sent packet.
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webrtc::ModuleRTPUtility::RTPHeaderParser rtp_parser(
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transport_.last_sent_packet_, rtp_length);
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webrtc::WebRtcRTPHeader rtp_header;
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RtpHeaderExtensionMap map;
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map.Register(kType, kId);
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const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
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ASSERT_TRUE(valid_rtp_header);
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// Verify transmission time offset.
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EXPECT_EQ(kStoredTimeInMs * 90, rtp_header.extension.transmissionTimeOffset);
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}
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TEST_F(RtpSenderTest, SendGenericVideo) {
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char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
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const uint8_t payload_type = 127;
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ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
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0, 1500));
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uint8_t payload[] = {47, 11, 32, 93, 89};
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// Send keyframe
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ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
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4321, payload, sizeof(payload),
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NULL));
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ModuleRTPUtility::RTPHeaderParser rtp_parser(transport_.last_sent_packet_,
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transport_.last_sent_packet_len_);
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webrtc::WebRtcRTPHeader rtp_header;
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ASSERT_TRUE(rtp_parser.Parse(rtp_header));
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const uint8_t* payload_data = ModuleRTPUtility::GetPayloadData(&rtp_header,
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transport_.last_sent_packet_);
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uint8_t generic_header = *payload_data++;
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ASSERT_EQ(sizeof(payload) + sizeof(generic_header),
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ModuleRTPUtility::GetPayloadDataLength(&rtp_header,
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transport_.last_sent_packet_len_));
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EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kKeyFrameBit);
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EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kFirstPacketBit);
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EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload)));
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// Send delta frame
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payload[0] = 13;
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payload[1] = 42;
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payload[4] = 13;
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ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
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1234, 4321, payload,
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sizeof(payload), NULL));
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ModuleRTPUtility::RTPHeaderParser rtp_parser2(transport_.last_sent_packet_,
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transport_.last_sent_packet_len_);
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ASSERT_TRUE(rtp_parser.Parse(rtp_header));
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payload_data = ModuleRTPUtility::GetPayloadData(&rtp_header,
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transport_.last_sent_packet_);
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generic_header = *payload_data++;
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EXPECT_FALSE(generic_header & RtpFormatVideoGeneric::kKeyFrameBit);
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EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kFirstPacketBit);
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ASSERT_EQ(sizeof(payload) + sizeof(generic_header),
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ModuleRTPUtility::GetPayloadDataLength(&rtp_header,
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transport_.last_sent_packet_len_));
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EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload)));
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}
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} // namespace webrtc
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