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webrtc/webrtc/video/call_perf_tests.cc
pbos@webrtc.org be9d2a4549 Reserve RTP/RTCP modules in SetSSRC. 
Allows setting SSRCs for future simulcast layers even though no set send
codec uses them.

Also re-enabling CanSwitchToUseAllSsrcs as an end-to-end test, required
for bitrate ramp-up, instead of send-side only (resolving issue 3078).
This test was used to verify reserved modules' SSRCs are preserved
correctly.

To enable a multiple-stream end-to-end test test::CallTest was modified
to work on a vector of receive streams instead of just one.

BUG=3078
R=kjellander@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15859005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6565 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-30 13:19:09 +00:00

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/*
* 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 <algorithm>
#include <sstream>
#include <string>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/call.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/rtp_to_ntp.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/thread_annotations.h"
#include "webrtc/test/call_test.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/test/encoder_settings.h"
#include "webrtc/test/fake_audio_device.h"
#include "webrtc/test/fake_decoder.h"
#include "webrtc/test/fake_encoder.h"
#include "webrtc/test/frame_generator.h"
#include "webrtc/test/frame_generator_capturer.h"
#include "webrtc/test/rtp_rtcp_observer.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/perf_test.h"
#include "webrtc/video/transport_adapter.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_codec.h"
#include "webrtc/voice_engine/include/voe_network.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#include "webrtc/voice_engine/include/voe_video_sync.h"
namespace webrtc {
class CallPerfTest : public test::CallTest {
protected:
void TestMinTransmitBitrate(bool pad_to_min_bitrate);
void TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
int threshold_ms,
int start_time_ms,
int run_time_ms);
};
class SyncRtcpObserver : public test::RtpRtcpObserver {
public:
explicit SyncRtcpObserver(const FakeNetworkPipe::Config& config)
: test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs, config),
crit_(CriticalSectionWrapper::CreateCriticalSection()) {}
virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
RTCPUtility::RTCPParserV2 parser(packet, length, true);
EXPECT_TRUE(parser.IsValid());
for (RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
packet_type != RTCPUtility::kRtcpNotValidCode;
packet_type = parser.Iterate()) {
if (packet_type == RTCPUtility::kRtcpSrCode) {
const RTCPUtility::RTCPPacket& packet = parser.Packet();
RtcpMeasurement ntp_rtp_pair(
packet.SR.NTPMostSignificant,
packet.SR.NTPLeastSignificant,
packet.SR.RTPTimestamp);
StoreNtpRtpPair(ntp_rtp_pair);
}
}
return SEND_PACKET;
}
int64_t RtpTimestampToNtp(uint32_t timestamp) const {
CriticalSectionScoped lock(crit_.get());
int64_t timestamp_in_ms = -1;
if (ntp_rtp_pairs_.size() == 2) {
// TODO(stefan): We can't EXPECT_TRUE on this call due to a bug in the
// RTCP sender where it sends RTCP SR before any RTP packets, which leads
// to a bogus NTP/RTP mapping.
RtpToNtpMs(timestamp, ntp_rtp_pairs_, &timestamp_in_ms);
return timestamp_in_ms;
}
return -1;
}
private:
void StoreNtpRtpPair(RtcpMeasurement ntp_rtp_pair) {
CriticalSectionScoped lock(crit_.get());
for (RtcpList::iterator it = ntp_rtp_pairs_.begin();
it != ntp_rtp_pairs_.end();
++it) {
if (ntp_rtp_pair.ntp_secs == it->ntp_secs &&
ntp_rtp_pair.ntp_frac == it->ntp_frac) {
// This RTCP has already been added to the list.
return;
}
}
// We need two RTCP SR reports to map between RTP and NTP. More than two
// will not improve the mapping.
if (ntp_rtp_pairs_.size() == 2) {
ntp_rtp_pairs_.pop_back();
}
ntp_rtp_pairs_.push_front(ntp_rtp_pair);
}
const scoped_ptr<CriticalSectionWrapper> crit_;
RtcpList ntp_rtp_pairs_ GUARDED_BY(crit_);
};
class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
static const int kInSyncThresholdMs = 50;
static const int kStartupTimeMs = 2000;
static const int kMinRunTimeMs = 30000;
public:
VideoRtcpAndSyncObserver(Clock* clock,
int voe_channel,
VoEVideoSync* voe_sync,
SyncRtcpObserver* audio_observer)
: SyncRtcpObserver(FakeNetworkPipe::Config()),
clock_(clock),
voe_channel_(voe_channel),
voe_sync_(voe_sync),
audio_observer_(audio_observer),
creation_time_ms_(clock_->TimeInMilliseconds()),
first_time_in_sync_(-1) {}
virtual void RenderFrame(const I420VideoFrame& video_frame,
int time_to_render_ms) OVERRIDE {
int64_t now_ms = clock_->TimeInMilliseconds();
uint32_t playout_timestamp = 0;
if (voe_sync_->GetPlayoutTimestamp(voe_channel_, playout_timestamp) != 0)
return;
int64_t latest_audio_ntp =
audio_observer_->RtpTimestampToNtp(playout_timestamp);
int64_t latest_video_ntp = RtpTimestampToNtp(video_frame.timestamp());
if (latest_audio_ntp < 0 || latest_video_ntp < 0)
return;
int time_until_render_ms =
std::max(0, static_cast<int>(video_frame.render_time_ms() - now_ms));
latest_video_ntp += time_until_render_ms;
int64_t stream_offset = latest_audio_ntp - latest_video_ntp;
std::stringstream ss;
ss << stream_offset;
webrtc::test::PrintResult("stream_offset",
"",
"synchronization",
ss.str(),
"ms",
false);
int64_t time_since_creation = now_ms - creation_time_ms_;
// During the first couple of seconds audio and video can falsely be
// estimated as being synchronized. We don't want to trigger on those.
if (time_since_creation < kStartupTimeMs)
return;
if (std::abs(latest_audio_ntp - latest_video_ntp) < kInSyncThresholdMs) {
if (first_time_in_sync_ == -1) {
first_time_in_sync_ = now_ms;
webrtc::test::PrintResult("sync_convergence_time",
"",
"synchronization",
time_since_creation,
"ms",
false);
}
if (time_since_creation > kMinRunTimeMs)
observation_complete_->Set();
}
}
private:
Clock* const clock_;
int voe_channel_;
VoEVideoSync* voe_sync_;
SyncRtcpObserver* audio_observer_;
int64_t creation_time_ms_;
int64_t first_time_in_sync_;
};
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSync) {
class AudioPacketReceiver : public PacketReceiver {
public:
AudioPacketReceiver(int channel, VoENetwork* voe_network)
: channel_(channel),
voe_network_(voe_network),
parser_(RtpHeaderParser::Create()) {}
virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
size_t length) OVERRIDE {
int ret;
if (parser_->IsRtcp(packet, static_cast<int>(length))) {
ret = voe_network_->ReceivedRTCPPacket(
channel_, packet, static_cast<unsigned int>(length));
} else {
ret = voe_network_->ReceivedRTPPacket(
channel_, packet, static_cast<unsigned int>(length), PacketTime());
}
return ret == 0 ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
}
private:
int channel_;
VoENetwork* voe_network_;
scoped_ptr<RtpHeaderParser> parser_;
};
VoiceEngine* voice_engine = VoiceEngine::Create();
VoEBase* voe_base = VoEBase::GetInterface(voice_engine);
VoECodec* voe_codec = VoECodec::GetInterface(voice_engine);
VoENetwork* voe_network = VoENetwork::GetInterface(voice_engine);
VoEVideoSync* voe_sync = VoEVideoSync::GetInterface(voice_engine);
const std::string audio_filename =
test::ResourcePath("voice_engine/audio_long16", "pcm");
ASSERT_STRNE("", audio_filename.c_str());
test::FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(),
audio_filename);
EXPECT_EQ(0, voe_base->Init(&fake_audio_device, NULL));
int channel = voe_base->CreateChannel();
FakeNetworkPipe::Config net_config;
net_config.queue_delay_ms = 500;
SyncRtcpObserver audio_observer(net_config);
VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(),
channel,
voe_sync,
&audio_observer);
Call::Config receiver_config(observer.ReceiveTransport());
receiver_config.voice_engine = voice_engine;
CreateCalls(Call::Config(observer.SendTransport()), receiver_config);
CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000};
EXPECT_EQ(0, voe_codec->SetSendCodec(channel, isac));
AudioPacketReceiver voe_packet_receiver(channel, voe_network);
audio_observer.SetReceivers(&voe_packet_receiver, &voe_packet_receiver);
internal::TransportAdapter transport_adapter(audio_observer.SendTransport());
transport_adapter.Enable();
EXPECT_EQ(0,
voe_network->RegisterExternalTransport(channel, transport_adapter));
observer.SetReceivers(receiver_call_->Receiver(), sender_call_->Receiver());
test::FakeDecoder fake_decoder;
CreateSendConfig(1);
CreateMatchingReceiveConfigs();
receive_configs_[0].renderer = &observer;
receive_configs_[0].audio_channel_id = channel;
CreateStreams();
CreateFrameGeneratorCapturer();
Start();
fake_audio_device.Start();
EXPECT_EQ(0, voe_base->StartPlayout(channel));
EXPECT_EQ(0, voe_base->StartReceive(channel));
EXPECT_EQ(0, voe_base->StartSend(channel));
EXPECT_EQ(kEventSignaled, observer.Wait())
<< "Timed out while waiting for audio and video to be synchronized.";
EXPECT_EQ(0, voe_base->StopSend(channel));
EXPECT_EQ(0, voe_base->StopReceive(channel));
EXPECT_EQ(0, voe_base->StopPlayout(channel));
fake_audio_device.Stop();
Stop();
observer.StopSending();
audio_observer.StopSending();
voe_base->DeleteChannel(channel);
voe_base->Release();
voe_codec->Release();
voe_network->Release();
voe_sync->Release();
DestroyStreams();
VoiceEngine::Delete(voice_engine);
}
void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
int threshold_ms,
int start_time_ms,
int run_time_ms) {
class CaptureNtpTimeObserver : public test::EndToEndTest,
public VideoRenderer {
public:
CaptureNtpTimeObserver(const FakeNetworkPipe::Config& config,
int threshold_ms,
int start_time_ms,
int run_time_ms)
: EndToEndTest(kLongTimeoutMs, config),
clock_(Clock::GetRealTimeClock()),
threshold_ms_(threshold_ms),
start_time_ms_(start_time_ms),
run_time_ms_(run_time_ms),
creation_time_ms_(clock_->TimeInMilliseconds()),
capturer_(NULL),
rtp_start_timestamp_set_(false),
rtp_start_timestamp_(0) {}
private:
virtual void RenderFrame(const I420VideoFrame& video_frame,
int time_to_render_ms) OVERRIDE {
if (video_frame.ntp_time_ms() <= 0) {
// Haven't got enough RTCP SR in order to calculate the capture ntp
// time.
return;
}
int64_t now_ms = clock_->TimeInMilliseconds();
int64_t time_since_creation = now_ms - creation_time_ms_;
if (time_since_creation < start_time_ms_) {
// Wait for |start_time_ms_| before start measuring.
return;
}
if (time_since_creation > run_time_ms_) {
observation_complete_->Set();
}
FrameCaptureTimeList::iterator iter =
capture_time_list_.find(video_frame.timestamp());
EXPECT_TRUE(iter != capture_time_list_.end());
// The real capture time has been wrapped to uint32_t before converted
// to rtp timestamp in the sender side. So here we convert the estimated
// capture time to a uint32_t 90k timestamp also for comparing.
uint32_t estimated_capture_timestamp =
90 * static_cast<uint32_t>(video_frame.ntp_time_ms());
uint32_t real_capture_timestamp = iter->second;
int time_offset_ms = real_capture_timestamp - estimated_capture_timestamp;
time_offset_ms = time_offset_ms / 90;
std::stringstream ss;
ss << time_offset_ms;
webrtc::test::PrintResult(
"capture_ntp_time", "", "real - estimated", ss.str(), "ms", true);
EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_);
}
virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, static_cast<int>(length), &header));
if (!rtp_start_timestamp_set_) {
// Calculate the rtp timestamp offset in order to calculate the real
// capture time.
uint32_t first_capture_timestamp =
90 * static_cast<uint32_t>(capturer_->first_frame_capture_time());
rtp_start_timestamp_ = header.timestamp - first_capture_timestamp;
rtp_start_timestamp_set_ = true;
}
uint32_t capture_timestamp = header.timestamp - rtp_start_timestamp_;
capture_time_list_.insert(
capture_time_list_.end(),
std::make_pair(header.timestamp, capture_timestamp));
return SEND_PACKET;
}
virtual void OnFrameGeneratorCapturerCreated(
test::FrameGeneratorCapturer* frame_generator_capturer) OVERRIDE {
capturer_ = frame_generator_capturer;
}
virtual void ModifyConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
std::vector<VideoStream>* video_streams) OVERRIDE {
(*receive_configs)[0].renderer = this;
// Enable the receiver side rtt calculation.
(*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true;
}
virtual void PerformTest() OVERRIDE {
EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for "
"estimated capture NTP time to be "
"within bounds.";
}
Clock* clock_;
int threshold_ms_;
int start_time_ms_;
int run_time_ms_;
int64_t creation_time_ms_;
test::FrameGeneratorCapturer* capturer_;
bool rtp_start_timestamp_set_;
uint32_t rtp_start_timestamp_;
typedef std::map<uint32_t, uint32_t> FrameCaptureTimeList;
FrameCaptureTimeList capture_time_list_;
} test(net_config, threshold_ms, start_time_ms, run_time_ms);
RunBaseTest(&test);
}
TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkDelay) {
FakeNetworkPipe::Config net_config;
net_config.queue_delay_ms = 100;
// TODO(wu): lower the threshold as the calculation/estimatation becomes more
// accurate.
const int kThresholdMs = 100;
const int kStartTimeMs = 10000;
const int kRunTimeMs = 20000;
TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
}
TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkJitter) {
FakeNetworkPipe::Config net_config;
net_config.queue_delay_ms = 100;
net_config.delay_standard_deviation_ms = 10;
// TODO(wu): lower the threshold as the calculation/estimatation becomes more
// accurate.
const int kThresholdMs = 100;
const int kStartTimeMs = 10000;
const int kRunTimeMs = 20000;
TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
}
TEST_F(CallPerfTest, RegisterCpuOveruseObserver) {
// Verifies that either a normal or overuse callback is triggered.
class OveruseCallbackObserver : public test::SendTest,
public webrtc::OveruseCallback {
public:
OveruseCallbackObserver() : SendTest(kLongTimeoutMs) {}
virtual void OnOveruse() OVERRIDE {
observation_complete_->Set();
}
virtual void OnNormalUse() OVERRIDE {
observation_complete_->Set();
}
virtual Call::Config GetSenderCallConfig() OVERRIDE {
Call::Config config(SendTransport());
config.overuse_callback = this;
return config;
}
virtual void PerformTest() OVERRIDE {
EXPECT_EQ(kEventSignaled, Wait())
<< "Timed out before receiving an overuse callback.";
}
} test;
RunBaseTest(&test);
}
void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {
static const int kMaxEncodeBitrateKbps = 30;
static const int kMinTransmitBitrateBps = 150000;
static const int kMinAcceptableTransmitBitrate = 130;
static const int kMaxAcceptableTransmitBitrate = 170;
static const int kNumBitrateObservationsInRange = 100;
class BitrateObserver : public test::EndToEndTest, public PacketReceiver {
public:
explicit BitrateObserver(bool using_min_transmit_bitrate)
: EndToEndTest(kLongTimeoutMs),
send_stream_(NULL),
send_transport_receiver_(NULL),
pad_to_min_bitrate_(using_min_transmit_bitrate),
num_bitrate_observations_in_range_(0) {}
private:
virtual void SetReceivers(PacketReceiver* send_transport_receiver,
PacketReceiver* receive_transport_receiver)
OVERRIDE {
send_transport_receiver_ = send_transport_receiver;
test::RtpRtcpObserver::SetReceivers(this, receive_transport_receiver);
}
virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
size_t length) OVERRIDE {
VideoSendStream::Stats stats = send_stream_->GetStats();
if (stats.substreams.size() > 0) {
assert(stats.substreams.size() == 1);
int bitrate_kbps = stats.substreams.begin()->second.bitrate_bps / 1000;
if (bitrate_kbps > 0) {
test::PrintResult(
"bitrate_stats_",
(pad_to_min_bitrate_ ? "min_transmit_bitrate"
: "without_min_transmit_bitrate"),
"bitrate_kbps",
static_cast<size_t>(bitrate_kbps),
"kbps",
false);
if (pad_to_min_bitrate_) {
if (bitrate_kbps > kMinAcceptableTransmitBitrate &&
bitrate_kbps < kMaxAcceptableTransmitBitrate) {
++num_bitrate_observations_in_range_;
}
} else {
// Expect bitrate stats to roughly match the max encode bitrate.
if (bitrate_kbps > kMaxEncodeBitrateKbps - 5 &&
bitrate_kbps < kMaxEncodeBitrateKbps + 5) {
++num_bitrate_observations_in_range_;
}
}
if (num_bitrate_observations_in_range_ ==
kNumBitrateObservationsInRange)
observation_complete_->Set();
}
}
return send_transport_receiver_->DeliverPacket(packet, length);
}
virtual void OnStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE {
send_stream_ = send_stream;
}
virtual void ModifyConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
std::vector<VideoStream>* video_streams) OVERRIDE {
if (pad_to_min_bitrate_) {
send_config->rtp.min_transmit_bitrate_bps = kMinTransmitBitrateBps;
} else {
assert(send_config->rtp.min_transmit_bitrate_bps == 0);
}
}
virtual void PerformTest() OVERRIDE {
EXPECT_EQ(kEventSignaled, Wait())
<< "Timeout while waiting for send-bitrate stats.";
}
VideoSendStream* send_stream_;
PacketReceiver* send_transport_receiver_;
const bool pad_to_min_bitrate_;
int num_bitrate_observations_in_range_;
} test(pad_to_min_bitrate);
fake_encoder_.SetMaxBitrate(kMaxEncodeBitrateKbps);
RunBaseTest(&test);
}
TEST_F(CallPerfTest, PadsToMinTransmitBitrate) { TestMinTransmitBitrate(true); }
TEST_F(CallPerfTest, NoPadWithoutMinTransmitBitrate) {
TestMinTransmitBitrate(false);
}
} // namespace webrtc
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