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Test that VideoSendStream responds to NACK.

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BUG=2228
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2194006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4715 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org 2013-09-10 14:56:33 +00:00
parent f880f863dd
commit df531a2eee
1 changed files with 94 additions and 2 deletions
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96
webrtc/video_engine/test/send_stream_tests.cc
View File

@ -9,9 +9,11 @@
*/
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "webrtc/video_engine/test/common/fake_encoder.h"
#include "webrtc/video_engine/test/common/frame_generator.h"
#include "webrtc/video_engine/test/common/frame_generator_capturer.h"
@ -67,6 +69,7 @@ class VideoSendStreamTest : public ::testing::Test {
VideoSendStream::Config config = call->GetDefaultSendConfig();
config.encoder = &fake_encoder_;
config.internal_source = false;
config.rtp.ssrcs.push_back(kSendSsrc);
test::FakeEncoder::SetCodecSettings(&config.codec, 1);
return config;
}
@ -97,7 +100,6 @@ TEST_F(VideoSendStreamTest, SendsSetSsrc) {
scoped_ptr<Call> call(Call::Create(call_config));
VideoSendStream::Config send_config = GetSendTestConfig(call.get());
send_config.rtp.ssrcs.push_back(kSendSsrc);
RunSendTest(call.get(), send_config, &observer);
}
@ -130,10 +132,100 @@ TEST_F(VideoSendStreamTest, SupportsCName) {
scoped_ptr<Call> call(Call::Create(call_config));
VideoSendStream::Config send_config = GetSendTestConfig(call.get());
send_config.rtp.ssrcs.push_back(kSendSsrc);
send_config.rtp.c_name = kCName;
RunSendTest(call.get(), send_config, &observer);
}
TEST_F(VideoSendStreamTest, RespondsToNack) {
class NackObserver : public SendTransportObserver, webrtc::Transport {
public:
NackObserver()
: SendTransportObserver(30 * 1000),
thread_(ThreadWrapper::CreateThread(NackProcess, this)),
send_call_receiver_(NULL),
send_count_(0),
ssrc_(0),
nacked_sequence_number_(0) {}
~NackObserver() {
EXPECT_TRUE(thread_->Stop());
}
void SetReceiver(PacketReceiver* send_call_receiver) {
send_call_receiver_ = send_call_receiver;
}
// Sending NACKs must be done from a different "network" thread to prevent
// violating locking orders. With this no locks are held prior to inserting
// packets back into the sender.
static bool NackProcess(void* observer) {
return static_cast<NackObserver*>(observer)->SendNack();
}
bool SendNack() {
NullReceiveStatistics null_stats;
RTCPSender rtcp_sender(0, false, Clock::GetRealTimeClock(), &null_stats);
EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(this));
rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
rtcp_sender.SetRemoteSSRC(ssrc_);
RTCPSender::FeedbackState feedback_state;
EXPECT_EQ(0, rtcp_sender.SendRTCP(
feedback_state, kRtcpNack, 1, &nacked_sequence_number_));
return false;
}
virtual int SendPacket(int channel, const void* data, int len) OVERRIDE {
ADD_FAILURE()
<< "This should never be reached. Only a NACK should be sent.";
return -1;
}
virtual int SendRTCPPacket(int channel,
const void* data,
int len) OVERRIDE {
EXPECT_TRUE(send_call_receiver_->DeliverPacket(
static_cast<const uint8_t*>(data), static_cast<size_t>(len)));
return len;
}
virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE {
EXPECT_TRUE(send_call_receiver_ != NULL);
RTPHeader header;
EXPECT_TRUE(
rtp_header_parser_->Parse(packet, static_cast<int>(length), &header));
// Nack second packet after receiving the third one.
if (++send_count_ == 3) {
ssrc_ = header.ssrc;
nacked_sequence_number_ = header.sequenceNumber - 1;
unsigned int id;
EXPECT_TRUE(thread_->Start(id));
}
if (header.sequenceNumber == nacked_sequence_number_)
send_test_complete_->Set();
return true;
}
private:
scoped_ptr<ThreadWrapper> thread_;
PacketReceiver* send_call_receiver_;
int send_count_;
uint32_t ssrc_;
uint16_t nacked_sequence_number_;
} observer;
Call::Config call_config(&observer);
scoped_ptr<Call> call(Call::Create(call_config));
observer.SetReceiver(call->Receiver());
VideoSendStream::Config send_config = GetSendTestConfig(call.get());
send_config.rtp.nack.rtp_history_ms = 1000;
RunSendTest(call.get(), send_config, &observer);
}
} // namespace webrtc