Files
platform-external-webrtc/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
stefan@webrtc.org 4ef438e2de Remove the send-side cname getter APIs from voice and video engine.
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.

R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 09:55:30 +00:00

155 lines
4.9 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 "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
#include <algorithm>
#include <vector>
using namespace webrtc;
class RtpRtcpAPITest : public ::testing::Test {
protected:
RtpRtcpAPITest() : module(NULL), fake_clock(123456) {
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);
rtp_payload_registry_.reset(new RTPPayloadRegistry(
RTPPayloadStrategy::CreateStrategy(true)));
rtp_receiver_.reset(RtpReceiver::CreateAudioReceiver(
test_id, &fake_clock, NULL, NULL, NULL, rtp_payload_registry_.get()));
}
virtual void TearDown() {
delete module;
}
int test_id;
scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
scoped_ptr<RtpReceiver> rtp_receiver_;
RtpRtcp* module;
uint32_t test_ssrc;
uint32_t test_timestamp;
uint16_t test_sequence_number;
uint32_t test_CSRC[webrtc::kRtpCsrcSize];
SimulatedClock 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) {
module->SetSSRC(test_ssrc);
EXPECT_EQ(test_ssrc, module->SSRC());
}
TEST_F(RtpRtcpAPITest, CSRC) {
EXPECT_EQ(0, module->SetCSRCs(test_CSRC, 2));
uint32_t 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"));
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, rtp_receiver_->NACK());
rtp_receiver_->SetNACKStatus(kNackRtcp);
EXPECT_EQ(kNackRtcp, rtp_receiver_->NACK());
}
TEST_F(RtpRtcpAPITest, RtxSender) {
unsigned int ssrc = 0;
int rtx_mode = kRtxOff;
const int kRtxPayloadType = 119;
int payload_type = -1;
module->SetRTXSendStatus(kRtxRetransmitted);
module->SetRtxSendPayloadType(kRtxPayloadType);
module->SetRtxSsrc(1);
module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
EXPECT_EQ(kRtxRetransmitted, rtx_mode);
EXPECT_EQ(1u, ssrc);
EXPECT_EQ(kRtxPayloadType, payload_type);
rtx_mode = kRtxOff;
module->SetRTXSendStatus(kRtxOff);
payload_type = -1;
module->SetRtxSendPayloadType(kRtxPayloadType);
module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
EXPECT_EQ(kRtxOff, rtx_mode);
EXPECT_EQ(kRtxPayloadType, payload_type);
module->SetRTXSendStatus(kRtxRetransmitted);
module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
EXPECT_EQ(kRtxRetransmitted, rtx_mode);
EXPECT_EQ(kRtxPayloadType, payload_type);
}
TEST_F(RtpRtcpAPITest, RtxReceiver) {
const uint32_t kRtxSsrc = 1;
const int kRtxPayloadType = 119;
EXPECT_FALSE(rtp_payload_registry_->RtxEnabled());
rtp_payload_registry_->SetRtxSsrc(kRtxSsrc);
rtp_payload_registry_->SetRtxPayloadType(kRtxPayloadType);
EXPECT_TRUE(rtp_payload_registry_->RtxEnabled());
RTPHeader rtx_header;
rtx_header.ssrc = kRtxSsrc;
rtx_header.payloadType = kRtxPayloadType;
EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
rtx_header.ssrc = 0;
EXPECT_FALSE(rtp_payload_registry_->IsRtx(rtx_header));
rtx_header.ssrc = kRtxSsrc;
rtx_header.payloadType = 0;
EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
}