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Add ramp-up tests for transport sequence number with and w/o audio.

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Also add a perf metric tracking the average network latency.

The audio stream test is disabled for now since audio isn't included in bitrate allocation.

BUG=webrtc:5263
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1582833002 .

Cr-Commit-Position: refs/heads/master@{#11244}
This commit is contained in:
Stefan Holmer
2016-01-14 10:00:21 +01:00
parent 709513d413
commit ff2a6351e0
6 changed files with 100 additions and 11 deletions
Download Patch File Download Diff File Expand all files Collapse all files

93
webrtc/call/rampup_tests.cc
View File

@ -55,6 +55,7 @@ RampUpTester::RampUpTester(size_t num_video_streams,
this,
"BitrateStatsPollingThread"),
sender_call_(nullptr) {
EXPECT_LE(num_audio_streams_, 1u);
if (rtx_) {
for (size_t i = 0; i < video_ssrcs_.size(); ++i)
rtx_ssrc_map_[video_rtx_ssrcs_[i]] = video_ssrcs_[i];
@ -91,6 +92,10 @@ size_t RampUpTester::GetNumVideoStreams() const {
return num_video_streams_;
}
size_t RampUpTester::GetNumAudioStreams() const {
return num_audio_streams_;
}
void RampUpTester::ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
@ -171,6 +176,37 @@ void RampUpTester::ModifyVideoConfigs(
}
}
void RampUpTester::ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs) {
if (num_audio_streams_ == 0)
return;
EXPECT_NE(RtpExtension::kTOffset, extension_type_)
<< "Audio BWE not supported with toffset.";
send_config->rtp.ssrc = audio_ssrcs_[0];
send_config->rtp.extensions.clear();
bool transport_cc = false;
if (extension_type_ == RtpExtension::kAbsSendTime) {
transport_cc = false;
send_config->rtp.extensions.push_back(
RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId));
} else if (extension_type_ == RtpExtension::kTransportSequenceNumber) {
transport_cc = true;
send_config->rtp.extensions.push_back(RtpExtension(
extension_type_.c_str(), kTransportSequenceNumberExtensionId));
}
for (AudioReceiveStream::Config& recv_config : *receive_configs) {
recv_config.combined_audio_video_bwe = true;
recv_config.rtp.transport_cc = transport_cc;
recv_config.rtp.extensions = send_config->rtp.extensions;
recv_config.rtp.remote_ssrc = send_config->rtp.ssrc;
}
}
void RampUpTester::OnCallsCreated(Call* sender_call, Call* receiver_call) {
sender_call_ = sender_call;
}
@ -231,6 +267,7 @@ void RampUpTester::AccumulateStats(const VideoSendStream::StreamStats& stream,
void RampUpTester::TriggerTestDone() {
RTC_DCHECK_GE(test_start_ms_, 0);
// TODO(holmer): Add audio send stats here too when those APIs are available.
VideoSendStream::Stats send_stats = send_stream_->GetStats();
size_t total_packets_sent = 0;
@ -264,6 +301,8 @@ void RampUpTester::TriggerTestDone() {
ReportResult("ramp-up-time", ramp_up_finished_ms_ - test_start_ms_,
"milliseconds");
}
ReportResult("ramp-up-average-network-latency",
send_transport_->GetAverageDelayMs(), "milliseconds");
}
void RampUpTester::PerformTest() {
@ -274,12 +313,18 @@ void RampUpTester::PerformTest() {
poller_thread_.Stop();
}
RampUpDownUpTester::RampUpDownUpTester(size_t num_streams,
RampUpDownUpTester::RampUpDownUpTester(size_t num_video_streams,
size_t num_audio_streams,
unsigned int start_bitrate_bps,
const std::string& extension_type,
bool rtx,
bool red)
: RampUpTester(num_streams, 0, start_bitrate_bps, extension_type, rtx, red),
: RampUpTester(num_video_streams,
num_audio_streams,
start_bitrate_bps,
extension_type,
rtx,
red),
test_state_(kFirstRampup),
state_start_ms_(clock_->TimeInMilliseconds()),
interval_start_ms_(clock_->TimeInMilliseconds()),
@ -375,6 +420,8 @@ void RampUpDownUpTester::EvolveTestState(int bitrate_bps, bool suspended) {
webrtc::test::PrintResult("ramp_up_down_up", GetModifierString(),
"second_rampup", now - state_start_ms_, "ms",
false);
ReportResult("ramp-up-down-up-average-network-latency",
send_transport_->GetAverageDelayMs(), "milliseconds");
observation_complete_.Set();
}
break;
@ -421,35 +468,59 @@ TEST_F(RampUpTest, SingleStreamWithHighStartBitrate) {
// Disabled on Mac due to flakiness, see
// https://bugs.chromium.org/p/webrtc/issues/detail?id=5407
#ifndef WEBRTC_MAC
static const uint32_t kStartBitrateBps = 60000;
TEST_F(RampUpTest, UpDownUpOneStream) {
RampUpDownUpTester test(1, 60000, RtpExtension::kAbsSendTime, false, false);
RampUpDownUpTester test(1, 0, kStartBitrateBps, RtpExtension::kAbsSendTime,
false, false);
RunBaseTest(&test);
}
TEST_F(RampUpTest, UpDownUpThreeStreams) {
RampUpDownUpTester test(3, 60000, RtpExtension::kAbsSendTime, false, false);
RampUpDownUpTester test(3, 0, kStartBitrateBps, RtpExtension::kAbsSendTime,
false, false);
RunBaseTest(&test);
}
TEST_F(RampUpTest, UpDownUpOneStreamRtx) {
RampUpDownUpTester test(1, 60000, RtpExtension::kAbsSendTime, true, false);
RampUpDownUpTester test(1, 0, kStartBitrateBps, RtpExtension::kAbsSendTime,
true, false);
RunBaseTest(&test);
}
TEST_F(RampUpTest, UpDownUpThreeStreamsRtx) {
RampUpDownUpTester test(3, 60000, RtpExtension::kAbsSendTime, true, false);
RampUpDownUpTester test(3, 0, kStartBitrateBps, RtpExtension::kAbsSendTime,
true, false);
RunBaseTest(&test);
}
TEST_F(RampUpTest, UpDownUpOneStreamByRedRtx) {
RampUpDownUpTester test(1, 60000, RtpExtension::kAbsSendTime, true, true);
RampUpDownUpTester test(1, 0, kStartBitrateBps, RtpExtension::kAbsSendTime,
true, true);
RunBaseTest(&test);
}
TEST_F(RampUpTest, UpDownUpThreeStreamsByRedRtx) {
RampUpDownUpTester test(3, 60000, RtpExtension::kAbsSendTime, true, true);
RampUpDownUpTester test(3, 0, kStartBitrateBps, RtpExtension::kAbsSendTime,
true, true);
RunBaseTest(&test);
}
TEST_F(RampUpTest, SendSideVideoUpDownUpRtx) {
RampUpDownUpTester test(3, 0, kStartBitrateBps,
RtpExtension::kTransportSequenceNumber, true, false);
RunBaseTest(&test);
}
// TODO(holmer): Enable when audio bitrates are included in the bitrate
// allocation.
TEST_F(RampUpTest, DISABLED_SendSideAudioVideoUpDownUpRtx) {
RampUpDownUpTester test(3, 1, kStartBitrateBps,
RtpExtension::kTransportSequenceNumber, true, false);
RunBaseTest(&test);
}
#endif
TEST_F(RampUpTest, AbsSendTimeSingleStream) {
@ -496,6 +567,12 @@ TEST_F(RampUpTest, TransportSequenceNumberSimulcastWithRtx) {
RunBaseTest(&test);
}
TEST_F(RampUpTest, AudioVideoTransportSequenceNumberSimulcastWithRtx) {
RampUpTester test(3, 1, 0, RtpExtension::kTransportSequenceNumber, true,
false);
RunBaseTest(&test);
}
TEST_F(RampUpTest, TransportSequenceNumberSimulcastByRedWithRtx) {
RampUpTester test(3, 0, 0, RtpExtension::kTransportSequenceNumber, true,
true);

7
webrtc/call/rampup_tests.h
View File

@ -40,6 +40,7 @@ class RampUpTester : public test::EndToEndTest {
~RampUpTester() override;
size_t GetNumVideoStreams() const override;
size_t GetNumAudioStreams() const override;
void PerformTest() override;
@ -79,6 +80,9 @@ class RampUpTester : public test::EndToEndTest {
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override;
void ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs) override;
void OnCallsCreated(Call* sender_call, Call* receiver_call) override;
static bool BitrateStatsPollingThread(void* obj);
@ -101,7 +105,8 @@ class RampUpTester : public test::EndToEndTest {
class RampUpDownUpTester : public RampUpTester {
public:
RampUpDownUpTester(size_t num_streams,
RampUpDownUpTester(size_t num_video_streams,
size_t num_audio_streams,
unsigned int start_bitrate_bps,
const std::string& extension_type,
bool rtx,

4
webrtc/test/direct_transport.cc
View File

@ -70,6 +70,10 @@ bool DirectTransport::SendRtcp(const uint8_t* data, size_t length) {
return true;
}
int DirectTransport::GetAverageDelayMs() {
return fake_network_.AverageDelay();
}
bool DirectTransport::NetworkProcess(void* transport) {
return static_cast<DirectTransport*>(transport)->SendPackets();
}

2
webrtc/test/direct_transport.h
View File

@ -46,6 +46,8 @@ class DirectTransport : public Transport {
const PacketOptions& options) override;
bool SendRtcp(const uint8_t* data, size_t length) override;
int GetAverageDelayMs();
private:
static bool NetworkProcess(void* transport);
bool SendPackets();

3
webrtc/test/fake_network_pipe.cc
View File

@ -146,7 +146,8 @@ int FakeNetworkPipe::AverageDelay() {
if (sent_packets_ == 0)
return 0;
return total_packet_delay_ / static_cast<int>(sent_packets_);
return static_cast<int>(total_packet_delay_ /
static_cast<int64_t>(sent_packets_));
}
void FakeNetworkPipe::Process() {

2
webrtc/test/fake_network_pipe.h
View File

@ -82,7 +82,7 @@ class FakeNetworkPipe {
// Statistics.
size_t dropped_packets_;
size_t sent_packets_;
int total_packet_delay_;
int64_t total_packet_delay_;
int64_t next_process_time_;