
Review URL: https://webrtc-codereview.appspot.com/934023 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3224 4adac7df-926f-26a2-2b94-8c16560cd09d
115 lines
3.2 KiB
C++
115 lines
3.2 KiB
C++
/*
|
|
* 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 <algorithm>
|
|
#include <vector>
|
|
#include <gtest/gtest.h>
|
|
|
|
#include "test_api.h"
|
|
|
|
#include "common_types.h"
|
|
#include "rtp_rtcp.h"
|
|
#include "rtp_rtcp_defines.h"
|
|
|
|
using namespace webrtc;
|
|
|
|
class RtpRtcpAPITest : public ::testing::Test {
|
|
protected:
|
|
RtpRtcpAPITest() {
|
|
test_CSRC[0] = 1234;
|
|
test_CSRC[1] = 2345;
|
|
test_id = 123;
|
|
test_ssrc = 3456;
|
|
test_timestamp = 4567;
|
|
test_sequence_number = 2345;
|
|
}
|
|
~RtpRtcpAPITest() {}
|
|
|
|
virtual void SetUp() {
|
|
RtpRtcp::Configuration configuration;
|
|
configuration.id = test_id;
|
|
configuration.audio = true;
|
|
configuration.clock = &fake_clock;
|
|
module = RtpRtcp::CreateRtpRtcp(configuration);
|
|
}
|
|
|
|
virtual void TearDown() {
|
|
delete module;
|
|
}
|
|
|
|
int test_id;
|
|
RtpRtcp* module;
|
|
WebRtc_UWord32 test_ssrc;
|
|
WebRtc_UWord32 test_timestamp;
|
|
WebRtc_UWord16 test_sequence_number;
|
|
WebRtc_UWord32 test_CSRC[webrtc::kRtpCsrcSize];
|
|
FakeRtpRtcpClock fake_clock;
|
|
};
|
|
|
|
TEST_F(RtpRtcpAPITest, Basic) {
|
|
EXPECT_EQ(0, module->SetSequenceNumber(test_sequence_number));
|
|
EXPECT_EQ(test_sequence_number, module->SequenceNumber());
|
|
|
|
EXPECT_EQ(0, module->SetStartTimestamp(test_timestamp));
|
|
EXPECT_EQ(test_timestamp, module->StartTimestamp());
|
|
|
|
EXPECT_FALSE(module->Sending());
|
|
EXPECT_EQ(0, module->SetSendingStatus(true));
|
|
EXPECT_TRUE(module->Sending());
|
|
}
|
|
|
|
TEST_F(RtpRtcpAPITest, MTU) {
|
|
EXPECT_EQ(-1, module->SetMaxTransferUnit(10));
|
|
EXPECT_EQ(-1, module->SetMaxTransferUnit(IP_PACKET_SIZE + 1));
|
|
EXPECT_EQ(0, module->SetMaxTransferUnit(1234));
|
|
EXPECT_EQ(1234-20-8, module->MaxPayloadLength());
|
|
|
|
EXPECT_EQ(0, module->SetTransportOverhead(true, true, 12));
|
|
EXPECT_EQ(1234 - 20- 20 -20 - 12, module->MaxPayloadLength());
|
|
|
|
EXPECT_EQ(0, module->SetTransportOverhead(false, false, 0));
|
|
EXPECT_EQ(1234 - 20 - 8, module->MaxPayloadLength());
|
|
}
|
|
|
|
TEST_F(RtpRtcpAPITest, SSRC) {
|
|
EXPECT_EQ(0, module->SetSSRC(test_ssrc));
|
|
EXPECT_EQ(test_ssrc, module->SSRC());
|
|
}
|
|
|
|
TEST_F(RtpRtcpAPITest, CSRC) {
|
|
EXPECT_EQ(0, module->SetCSRCs(test_CSRC, 2));
|
|
WebRtc_UWord32 testOfCSRC[webrtc::kRtpCsrcSize];
|
|
EXPECT_EQ(2, module->CSRCs(testOfCSRC));
|
|
EXPECT_EQ(test_CSRC[0], testOfCSRC[0]);
|
|
EXPECT_EQ(test_CSRC[1], testOfCSRC[1]);
|
|
}
|
|
|
|
TEST_F(RtpRtcpAPITest, RTCP) {
|
|
EXPECT_EQ(kRtcpOff, module->RTCP());
|
|
EXPECT_EQ(0, module->SetRTCPStatus(kRtcpCompound));
|
|
EXPECT_EQ(kRtcpCompound, module->RTCP());
|
|
|
|
EXPECT_EQ(0, module->SetCNAME("john.doe@test.test"));
|
|
|
|
char cName[RTCP_CNAME_SIZE];
|
|
EXPECT_EQ(0, module->CNAME(cName));
|
|
EXPECT_STRCASEEQ(cName, "john.doe@test.test");
|
|
|
|
EXPECT_FALSE(module->TMMBR());
|
|
EXPECT_EQ(0, module->SetTMMBRStatus(true));
|
|
EXPECT_TRUE(module->TMMBR());
|
|
EXPECT_EQ(0, module->SetTMMBRStatus(false));
|
|
EXPECT_FALSE(module->TMMBR());
|
|
|
|
EXPECT_EQ(kNackOff, module->NACK());
|
|
EXPECT_EQ(0, module->SetNACKStatus(kNackRtcp));
|
|
EXPECT_EQ(kNackRtcp, module->NACK());
|
|
}
|