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// Copyright 2018 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "base/command_line.h"
#include "base/strings/stringprintf.h"
#include "base/test/test_timeouts.h"
#include "base/time/time.h"
#include "chrome/browser/media/webrtc/test_stats_dictionary.h"
#include "chrome/browser/media/webrtc/webrtc_browsertest_base.h"
#include "chrome/browser/media/webrtc/webrtc_browsertest_common.h"
#include "content/public/browser/web_contents.h"
#include "content/public/common/content_switches.h"
#include "media/base/media_switches.h"
#include "testing/perf/perf_result_reporter.h"
#include "ui/gl/gl_switches.h"
namespace {
static const char kMainWebrtcTestHtmlPage[] =
"/webrtc/webrtc_video_display_perf_test.html";
static const char kInboundRtp[] = "inbound-rtp";
static const char kOutboundRtp[] = "outbound-rtp";
constexpr int kBitsPerByte = 8;
constexpr char kMetricPrefixHighBitrate[] = "WebRtcHighBitrateVideo.";
constexpr char kMetricSendRateBitsPerS[] = "send_rate";
constexpr char kMetricReceiveRateBitsPerS[] = "receive_rate";
perf_test::PerfResultReporter SetUpReporter(const std::string& story) {
perf_test::PerfResultReporter reporter(kMetricPrefixHighBitrate, story);
reporter.RegisterFyiMetric(kMetricSendRateBitsPerS, "bits/s");
reporter.RegisterFyiMetric(kMetricReceiveRateBitsPerS, "bits/s");
return reporter;
}
// Sums up "RTC[In/Out]boundRTPStreamStats.bytes_[received/sent]" values.
double GetTotalRTPStreamBytes(content::TestStatsReportDictionary* report,
const char* type,
const char* media_type) {
DCHECK(type == kInboundRtp || type == kOutboundRtp);
const char* bytes_name =
(type == kInboundRtp) ? "bytesReceived" : "bytesSent";
double total_bytes = 0.0;
report->ForEach([&type, &bytes_name, &media_type,
&total_bytes](const content::TestStatsDictionary& stats) {
if (stats.GetString("type") == type &&
stats.GetString("mediaType") == media_type) {
total_bytes += stats.GetNumber(bytes_name);
}
});
return total_bytes;
}
double GetVideoBytesSent(content::TestStatsReportDictionary* report) {
return GetTotalRTPStreamBytes(report, kOutboundRtp, "video");
}
double GetVideoBytesReceived(content::TestStatsReportDictionary* report) {
return GetTotalRTPStreamBytes(report, kInboundRtp, "video");
}
} // anonymous namespace
namespace content {
// Tests the performance of WebRTC peer connection with high bitrate
//
// This test creates a WebRTC peer connection between two tabs and sets a very
// high target bitrate to observe any perf regressions/improvements for such
// cases. In order to achieve this, we use a fake codec that creates a dummy
// output for the given bitrate.
class WebRtcVideoHighBitrateBrowserTest : public WebRtcTestBase {
public:
void SetUpInProcessBrowserTestFixture() override {
DetectErrorsInJavaScript();
}
void SetUpCommandLine(base::CommandLine* command_line) override {
command_line->AppendSwitch(switches::kUseFakeCodecForPeerConnection);
command_line->AppendSwitch(switches::kUseFakeDeviceForMediaStream);
command_line->AppendSwitch(switches::kUseFakeUIForMediaStream);
command_line->AppendSwitch(switches::kUseGpuInTests);
}
protected:
void SetDefaultVideoTargetBitrate(content::WebContents* tab,
int bits_per_second) {
EXPECT_EQ("ok", ExecuteJavascript(
base::StringPrintf("setDefaultVideoTargetBitrate(%d)",
bits_per_second),
tab));
}
};
IN_PROC_BROWSER_TEST_F(WebRtcVideoHighBitrateBrowserTest,
MANUAL_HighBitrateEncodeDecode) {
ASSERT_TRUE(embedded_test_server()->Start());
ASSERT_GE(TestTimeouts::test_launcher_timeout().InSeconds(), 30)
<< "This is a long-running test; you must specify "
"--test-launcher-timeout to have a value of at least 30000.";
ASSERT_GE(TestTimeouts::action_max_timeout().InSeconds(), 30)
<< "This is a long-running test; you must specify "
"--ui-test-action-max-timeout to have a value of at least 30000.";
ASSERT_LT(TestTimeouts::action_max_timeout(),
TestTimeouts::test_launcher_timeout())
<< "action_max_timeout needs to be strictly-less-than "
"test_launcher_timeout";
content::WebContents* left_tab =
OpenPageAndGetUserMediaInNewTabWithConstraints(
embedded_test_server()->GetURL(kMainWebrtcTestHtmlPage),
"{audio: true, video: true}");
content::WebContents* right_tab =
OpenPageAndGetUserMediaInNewTabWithConstraints(
embedded_test_server()->GetURL(kMainWebrtcTestHtmlPage),
"{audio: true, video: false}");
SetupPeerconnectionWithLocalStream(left_tab);
SetupPeerconnectionWithLocalStream(right_tab);
const int target_bits_per_second = 80000;
SetDefaultVideoTargetBitrate(left_tab, target_bits_per_second);
SetDefaultVideoTargetBitrate(right_tab, target_bits_per_second);
NegotiateCall(left_tab, right_tab);
// Run the connection a bit to ramp up.
test::SleepInJavascript(left_tab, 10000);
scoped_refptr<TestStatsReportDictionary> sender_report =
GetStatsReportDictionary(left_tab);
const double video_bytes_sent_before = GetVideoBytesSent(sender_report.get());
scoped_refptr<TestStatsReportDictionary> receiver_report =
GetStatsReportDictionary(right_tab);
const double video_bytes_received_before =
GetVideoBytesReceived(receiver_report.get());
// Collect stats.
const double duration_in_seconds = 5.0;
test::SleepInJavascript(
left_tab, duration_in_seconds * base::Time::kMillisecondsPerSecond);
sender_report = GetStatsReportDictionary(left_tab);
const double video_bytes_sent_after = GetVideoBytesSent(sender_report.get());
receiver_report = GetStatsReportDictionary(right_tab);
const double video_bytes_received_after =
GetVideoBytesReceived(receiver_report.get());
const double video_send_rate =
(video_bytes_sent_after - video_bytes_sent_before) / duration_in_seconds;
const double video_receive_rate =
(video_bytes_received_after - video_bytes_received_before) /
duration_in_seconds;
auto reporter = SetUpReporter("baseline_story");
reporter.AddResult(kMetricSendRateBitsPerS, video_send_rate * kBitsPerByte);
reporter.AddResult(kMetricReceiveRateBitsPerS,
video_receive_rate * kBitsPerByte);
HangUp(left_tab);
HangUp(right_tab);
}
} // namespace content
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