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Split up EngineTests and RampupTests.

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This allows having one group of tests per file, the test files are
long enough as they are.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2196004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4712 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org 2013-09-10 09:26:25 +00:00
parent eda189be14
commit 744fbc7fe4
3 changed files with 190 additions and 158 deletions
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159
webrtc/video_engine/test/engine_tests.cc
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@ -13,14 +13,11 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/video_engine/new_include/call.h"
#include "webrtc/video_engine/test/common/direct_transport.h"
#include "webrtc/video_engine/test/common/fake_decoder.h"
@ -32,160 +29,6 @@
namespace webrtc {
class StreamObserver : public newapi::Transport, public RemoteBitrateObserver {
public:
typedef std::map<uint32_t, int> BytesSentMap;
StreamObserver(int num_expected_ssrcs,
newapi::Transport* feedback_transport,
Clock* clock)
: critical_section_(CriticalSectionWrapper::CreateCriticalSection()),
all_ssrcs_sent_(EventWrapper::Create()),
rtp_parser_(RtpHeaderParser::Create()),
feedback_transport_(new TransportWrapper(feedback_transport)),
receive_stats_(ReceiveStatistics::Create(clock)),
clock_(clock),
num_expected_ssrcs_(num_expected_ssrcs) {
// Ideally we would only have to instantiate an RtcpSender, an
// RtpHeaderParser and a RemoteBitrateEstimator here, but due to the current
// state of the RTP module we need a full module and receive statistics to
// be able to produce an RTCP with REMB.
RtpRtcp::Configuration config;
config.receive_statistics = receive_stats_.get();
config.outgoing_transport = feedback_transport_.get();
rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
rtp_rtcp_->SetREMBStatus(true);
rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound);
rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
1);
AbsoluteSendTimeRemoteBitrateEstimatorFactory rbe_factory;
remote_bitrate_estimator_.reset(rbe_factory.Create(this, clock));
}
virtual void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
unsigned int bitrate) {
CriticalSectionScoped lock(critical_section_.get());
if (ssrcs.size() == num_expected_ssrcs_ && bitrate >= kExpectedBitrateBps)
all_ssrcs_sent_->Set();
rtp_rtcp_->SetREMBData(
bitrate, static_cast<uint8_t>(ssrcs.size()), &ssrcs[0]);
rtp_rtcp_->Process();
}
virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE {
CriticalSectionScoped lock(critical_section_.get());
RTPHeader header;
EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header));
receive_stats_->IncomingPacket(header, length, false);
rtp_rtcp_->SetRemoteSSRC(header.ssrc);
remote_bitrate_estimator_->IncomingPacket(
clock_->TimeInMilliseconds(), static_cast<int>(length - 12), header);
if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) {
remote_bitrate_estimator_->Process();
}
return true;
}
virtual bool SendRTCP(const uint8_t* packet, size_t length) OVERRIDE {
return true;
}
EventTypeWrapper Wait() { return all_ssrcs_sent_->Wait(120 * 1000); }
private:
class TransportWrapper : public webrtc::Transport {
public:
explicit TransportWrapper(newapi::Transport* new_transport)
: new_transport_(new_transport) {}
virtual int SendPacket(int channel, const void* data, int len) OVERRIDE {
return new_transport_->SendRTP(static_cast<const uint8_t*>(data), len)
? len
: -1;
}
virtual int SendRTCPPacket(int channel,
const void* data,
int len) OVERRIDE {
return new_transport_->SendRTCP(static_cast<const uint8_t*>(data), len)
? len
: -1;
}
private:
newapi::Transport* new_transport_;
};
static const unsigned int kExpectedBitrateBps = 1200000;
scoped_ptr<CriticalSectionWrapper> critical_section_;
scoped_ptr<EventWrapper> all_ssrcs_sent_;
scoped_ptr<RtpHeaderParser> rtp_parser_;
scoped_ptr<RtpRtcp> rtp_rtcp_;
scoped_ptr<TransportWrapper> feedback_transport_;
scoped_ptr<ReceiveStatistics> receive_stats_;
scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
Clock* clock_;
const size_t num_expected_ssrcs_;
};
class RampUpTest : public ::testing::TestWithParam<bool> {
public:
virtual void SetUp() { reserved_ssrcs_.clear(); }
protected:
std::map<uint32_t, bool> reserved_ssrcs_;
};
TEST_P(RampUpTest, RampUpWithPadding) {
test::DirectTransport receiver_transport;
StreamObserver stream_observer(
3, &receiver_transport, Clock::GetRealTimeClock());
Call::Config call_config(&stream_observer);
scoped_ptr<Call> call(Call::Create(call_config));
VideoSendStream::Config send_config = call->GetDefaultSendConfig();
receiver_transport.SetReceiver(call->Receiver());
test::FakeEncoder encoder(Clock::GetRealTimeClock());
send_config.encoder = &encoder;
send_config.internal_source = false;
test::FakeEncoder::SetCodecSettings(&send_config.codec, 3);
send_config.pacing = GetParam();
test::GenerateRandomSsrcs(&send_config, &reserved_ssrcs_);
VideoSendStream* send_stream = call->CreateSendStream(send_config);
VideoReceiveStream::Config receive_config;
receive_config.rtp.ssrc = send_config.rtp.ssrcs[0];
receive_config.rtp.nack.rtp_history_ms = send_config.rtp.nack.rtp_history_ms;
VideoReceiveStream* receive_stream =
call->CreateReceiveStream(receive_config);
scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
test::FrameGeneratorCapturer::Create(
send_stream->Input(),
test::FrameGenerator::Create(send_config.codec.width,
send_config.codec.height,
Clock::GetRealTimeClock()),
30));
receive_stream->StartReceive();
send_stream->StartSend();
frame_generator_capturer->Start();
EXPECT_EQ(kEventSignaled, stream_observer.Wait());
frame_generator_capturer->Stop();
send_stream->StopSend();
receive_stream->StopReceive();
call->DestroyReceiveStream(receive_stream);
call->DestroySendStream(send_stream);
}
INSTANTIATE_TEST_CASE_P(RampUpTest, RampUpTest, ::testing::Bool());
struct EngineTestParams {
size_t width, height;
struct {

188
webrtc/video_engine/test/rampup_tests.cc Normal file
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@ -0,0 +1,188 @@
/*
* Copyright (c) 2013 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 <assert.h>
#include <map>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/video_engine/new_include/call.h"
#include "webrtc/video_engine/test/common/direct_transport.h"
#include "webrtc/video_engine/test/common/fake_decoder.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"
#include "webrtc/video_engine/test/common/generate_ssrcs.h"
namespace webrtc {
class StreamObserver : public newapi::Transport, public RemoteBitrateObserver {
public:
typedef std::map<uint32_t, int> BytesSentMap;
StreamObserver(int num_expected_ssrcs,
newapi::Transport* feedback_transport,
Clock* clock)
: critical_section_(CriticalSectionWrapper::CreateCriticalSection()),
all_ssrcs_sent_(EventWrapper::Create()),
rtp_parser_(RtpHeaderParser::Create()),
feedback_transport_(new TransportWrapper(feedback_transport)),
receive_stats_(ReceiveStatistics::Create(clock)),
clock_(clock),
num_expected_ssrcs_(num_expected_ssrcs) {
// Ideally we would only have to instantiate an RtcpSender, an
// RtpHeaderParser and a RemoteBitrateEstimator here, but due to the current
// state of the RTP module we need a full module and receive statistics to
// be able to produce an RTCP with REMB.
RtpRtcp::Configuration config;
config.receive_statistics = receive_stats_.get();
config.outgoing_transport = feedback_transport_.get();
rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
rtp_rtcp_->SetREMBStatus(true);
rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound);
rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
1);
AbsoluteSendTimeRemoteBitrateEstimatorFactory rbe_factory;
remote_bitrate_estimator_.reset(rbe_factory.Create(this, clock));
}
virtual void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
unsigned int bitrate) {
CriticalSectionScoped lock(critical_section_.get());
if (ssrcs.size() == num_expected_ssrcs_ && bitrate >= kExpectedBitrateBps)
all_ssrcs_sent_->Set();
rtp_rtcp_->SetREMBData(
bitrate, static_cast<uint8_t>(ssrcs.size()), &ssrcs[0]);
rtp_rtcp_->Process();
}
virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE {
CriticalSectionScoped lock(critical_section_.get());
RTPHeader header;
EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header));
receive_stats_->IncomingPacket(header, length, false);
rtp_rtcp_->SetRemoteSSRC(header.ssrc);
remote_bitrate_estimator_->IncomingPacket(
clock_->TimeInMilliseconds(), static_cast<int>(length - 12), header);
if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) {
remote_bitrate_estimator_->Process();
}
return true;
}
virtual bool SendRTCP(const uint8_t* packet, size_t length) OVERRIDE {
return true;
}
EventTypeWrapper Wait() { return all_ssrcs_sent_->Wait(120 * 1000); }
private:
class TransportWrapper : public webrtc::Transport {
public:
explicit TransportWrapper(newapi::Transport* new_transport)
: new_transport_(new_transport) {}
virtual int SendPacket(int channel, const void* data, int len) OVERRIDE {
return new_transport_->SendRTP(static_cast<const uint8_t*>(data), len)
? len
: -1;
}
virtual int SendRTCPPacket(int channel,
const void* data,
int len) OVERRIDE {
return new_transport_->SendRTCP(static_cast<const uint8_t*>(data), len)
? len
: -1;
}
private:
newapi::Transport* new_transport_;
};
static const unsigned int kExpectedBitrateBps = 1200000;
scoped_ptr<CriticalSectionWrapper> critical_section_;
scoped_ptr<EventWrapper> all_ssrcs_sent_;
scoped_ptr<RtpHeaderParser> rtp_parser_;
scoped_ptr<RtpRtcp> rtp_rtcp_;
scoped_ptr<TransportWrapper> feedback_transport_;
scoped_ptr<ReceiveStatistics> receive_stats_;
scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
Clock* clock_;
const size_t num_expected_ssrcs_;
};
class RampUpTest : public ::testing::TestWithParam<bool> {
public:
virtual void SetUp() { reserved_ssrcs_.clear(); }
protected:
std::map<uint32_t, bool> reserved_ssrcs_;
};
TEST_P(RampUpTest, RampUpWithPadding) {
test::DirectTransport receiver_transport;
StreamObserver stream_observer(
3, &receiver_transport, Clock::GetRealTimeClock());
Call::Config call_config(&stream_observer);
scoped_ptr<Call> call(Call::Create(call_config));
VideoSendStream::Config send_config = call->GetDefaultSendConfig();
receiver_transport.SetReceiver(call->Receiver());
test::FakeEncoder encoder(Clock::GetRealTimeClock());
send_config.encoder = &encoder;
send_config.internal_source = false;
test::FakeEncoder::SetCodecSettings(&send_config.codec, 3);
send_config.pacing = GetParam();
test::GenerateRandomSsrcs(&send_config, &reserved_ssrcs_);
VideoSendStream* send_stream = call->CreateSendStream(send_config);
VideoReceiveStream::Config receive_config;
receive_config.rtp.ssrc = send_config.rtp.ssrcs[0];
receive_config.rtp.nack.rtp_history_ms = send_config.rtp.nack.rtp_history_ms;
VideoReceiveStream* receive_stream =
call->CreateReceiveStream(receive_config);
scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
test::FrameGeneratorCapturer::Create(
send_stream->Input(),
test::FrameGenerator::Create(send_config.codec.width,
send_config.codec.height,
Clock::GetRealTimeClock()),
30));
receive_stream->StartReceive();
send_stream->StartSend();
frame_generator_capturer->Start();
EXPECT_EQ(kEventSignaled, stream_observer.Wait());
frame_generator_capturer->Stop();
send_stream->StopSend();
receive_stream->StopReceive();
call->DestroyReceiveStream(receive_stream);
call->DestroySendStream(send_stream);
}
INSTANTIATE_TEST_CASE_P(RampUpTest, RampUpTest, ::testing::Bool());
} // namespace webrtc

1
webrtc/video_engine/test/tests.gypi
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@ -139,6 +139,7 @@
'sources': [
'engine_tests.cc',
'full_stack.cc',
'rampup_tests.cc',
'send_stream_tests.cc',
'test_main.cc',
],