Add ramp-up tests for transport sequence number with and w/o audio.
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}
diff --git a/webrtc/call/rampup_tests.cc b/webrtc/call/rampup_tests.cc
index a3fcc30..81f1e81 100644
--- a/webrtc/call/rampup_tests.cc
+++ b/webrtc/call/rampup_tests.cc
@@ -55,6 +55,7 @@
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 @@
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::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::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 @@
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 @@
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 @@
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 @@
// 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 @@
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);
diff --git a/webrtc/call/rampup_tests.h b/webrtc/call/rampup_tests.h
index 89e7075..31a0a02 100644
--- a/webrtc/call/rampup_tests.h
+++ b/webrtc/call/rampup_tests.h
@@ -40,6 +40,7 @@
~RampUpTester() override;
size_t GetNumVideoStreams() const override;
+ size_t GetNumAudioStreams() const override;
void PerformTest() override;
@@ -79,6 +80,9 @@
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 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,
diff --git a/webrtc/test/direct_transport.cc b/webrtc/test/direct_transport.cc
index 6f7f998..591e154 100644
--- a/webrtc/test/direct_transport.cc
+++ b/webrtc/test/direct_transport.cc
@@ -70,6 +70,10 @@
return true;
}
+int DirectTransport::GetAverageDelayMs() {
+ return fake_network_.AverageDelay();
+}
+
bool DirectTransport::NetworkProcess(void* transport) {
return static_cast<DirectTransport*>(transport)->SendPackets();
}
diff --git a/webrtc/test/direct_transport.h b/webrtc/test/direct_transport.h
index 444ab26..d68bc71 100644
--- a/webrtc/test/direct_transport.h
+++ b/webrtc/test/direct_transport.h
@@ -46,6 +46,8 @@
const PacketOptions& options) override;
bool SendRtcp(const uint8_t* data, size_t length) override;
+ int GetAverageDelayMs();
+
private:
static bool NetworkProcess(void* transport);
bool SendPackets();
diff --git a/webrtc/test/fake_network_pipe.cc b/webrtc/test/fake_network_pipe.cc
index 4e43122..491a052 100644
--- a/webrtc/test/fake_network_pipe.cc
+++ b/webrtc/test/fake_network_pipe.cc
@@ -146,7 +146,8 @@
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() {
diff --git a/webrtc/test/fake_network_pipe.h b/webrtc/test/fake_network_pipe.h
index 1af63b0..5d589d8 100644
--- a/webrtc/test/fake_network_pipe.h
+++ b/webrtc/test/fake_network_pipe.h
@@ -82,7 +82,7 @@
// Statistics.
size_t dropped_packets_;
size_t sent_packets_;
- int total_packet_delay_;
+ int64_t total_packet_delay_;
int64_t next_process_time_;
