blob: e2afa4285680effd6c583968e2e23d7018b0e76d [file] [log] [blame]
/*
* 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 "modules/rtp_rtcp/test/testAPI/test_api.h"
#include <algorithm>
#include <memory>
#include <vector>
#include "rtc_base/checks.h"
#include "rtc_base/rate_limiter.h"
#include "test/null_transport.h"
namespace webrtc {
void LoopBackTransport::SetSendModule(RtpRtcp* rtp_rtcp_module,
RTPPayloadRegistry* payload_registry,
RtpReceiver* receiver,
ReceiveStatistics* receive_statistics) {
rtp_rtcp_module_ = rtp_rtcp_module;
rtp_payload_registry_ = payload_registry;
rtp_receiver_ = receiver;
receive_statistics_ = receive_statistics;
}
void LoopBackTransport::DropEveryNthPacket(int n) {
packet_loss_ = n;
}
bool LoopBackTransport::SendRtp(const uint8_t* data,
size_t len,
const PacketOptions& options) {
count_++;
if (packet_loss_ > 0) {
if ((count_ % packet_loss_) == 0) {
return true;
}
}
RTPHeader header;
std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
if (!parser->Parse(data, len, &header)) {
return false;
}
const auto pl =
rtp_payload_registry_->PayloadTypeToPayload(header.payloadType);
if (!pl) {
return false;
}
const uint8_t* payload = data + header.headerLength;
RTC_CHECK_GE(len, header.headerLength);
const size_t payload_length = len - header.headerLength;
receive_statistics_->IncomingPacket(header, len, false);
return rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
pl->typeSpecific);
}
bool LoopBackTransport::SendRtcp(const uint8_t* data, size_t len) {
rtp_rtcp_module_->IncomingRtcpPacket((const uint8_t*)data, len);
return true;
}
int32_t TestRtpReceiver::OnReceivedPayloadData(
const uint8_t* payload_data,
size_t payload_size,
const webrtc::WebRtcRTPHeader* rtp_header) {
EXPECT_LE(payload_size, sizeof(payload_data_));
memcpy(payload_data_, payload_data, payload_size);
memcpy(&rtp_header_, rtp_header, sizeof(rtp_header_));
payload_size_ = payload_size;
return 0;
}
class RtpRtcpAPITest : public ::testing::Test {
protected:
RtpRtcpAPITest()
: fake_clock_(123456), retransmission_rate_limiter_(&fake_clock_, 1000) {
test_csrcs_.push_back(1234);
test_csrcs_.push_back(2345);
test_ssrc_ = 3456;
test_timestamp_ = 4567;
test_sequence_number_ = 2345;
}
~RtpRtcpAPITest() {}
const uint32_t initial_ssrc = 8888;
void SetUp() override {
RtpRtcp::Configuration configuration;
configuration.audio = true;
configuration.clock = &fake_clock_;
configuration.outgoing_transport = &null_transport_;
configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
module_.reset(RtpRtcp::CreateRtpRtcp(configuration));
module_->SetSSRC(initial_ssrc);
rtp_payload_registry_.reset(new RTPPayloadRegistry());
}
std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
std::unique_ptr<RtpRtcp> module_;
uint32_t test_ssrc_;
uint32_t test_timestamp_;
uint16_t test_sequence_number_;
std::vector<uint32_t> test_csrcs_;
SimulatedClock fake_clock_;
test::NullTransport null_transport_;
RateLimiter retransmission_rate_limiter_;
};
TEST_F(RtpRtcpAPITest, Basic) {
module_->SetSequenceNumber(test_sequence_number_);
EXPECT_EQ(test_sequence_number_, module_->SequenceNumber());
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, PacketSize) {
module_->SetMaxRtpPacketSize(1234);
EXPECT_EQ(1234u, module_->MaxRtpPacketSize());
}
TEST_F(RtpRtcpAPITest, SSRC) {
module_->SetSSRC(test_ssrc_);
EXPECT_EQ(test_ssrc_, module_->SSRC());
}
TEST_F(RtpRtcpAPITest, RTCP) {
EXPECT_EQ(RtcpMode::kOff, module_->RTCP());
module_->SetRTCPStatus(RtcpMode::kCompound);
EXPECT_EQ(RtcpMode::kCompound, module_->RTCP());
EXPECT_EQ(0, module_->SetCNAME("john.doe@test.test"));
EXPECT_FALSE(module_->TMMBR());
module_->SetTMMBRStatus(true);
EXPECT_TRUE(module_->TMMBR());
module_->SetTMMBRStatus(false);
EXPECT_FALSE(module_->TMMBR());
}
TEST_F(RtpRtcpAPITest, RtxSender) {
module_->SetRtxSendStatus(kRtxRetransmitted);
EXPECT_EQ(kRtxRetransmitted, module_->RtxSendStatus());
module_->SetRtxSendStatus(kRtxOff);
EXPECT_EQ(kRtxOff, module_->RtxSendStatus());
module_->SetRtxSendStatus(kRtxRetransmitted);
EXPECT_EQ(kRtxRetransmitted, module_->RtxSendStatus());
}
} // namespace webrtc