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// Copyright 2014 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/files/file_util.h"
#include "base/json/json_reader.h"
#include "base/macros.h"
#include "base/memory/scoped_ptr.h"
#include "base/strings/string_split.h"
#include "base/strings/stringprintf.h"
#include "base/test/test_timeouts.h"
#include "base/time/time.h"
#include "chrome/browser/browser_process.h"
#include "chrome/browser/media/webrtc_browsertest_base.h"
#include "chrome/browser/media/webrtc_browsertest_common.h"
#include "chrome/browser/media/webrtc_browsertest_perf.h"
#include "chrome/browser/ui/browser.h"
#include "chrome/browser/ui/browser_tabstrip.h"
#include "chrome/browser/ui/tabs/tab_strip_model.h"
#include "chrome/common/chrome_switches.h"
#include "chrome/test/base/in_process_browser_test.h"
#include "chrome/test/base/ui_test_utils.h"
#include "content/public/common/content_switches.h"
#include "content/public/common/feature_h264_with_openh264_ffmpeg.h"
#include "content/public/common/features.h"
#include "content/public/test/browser_test_utils.h"
#include "media/base/media_switches.h"
#include "net/test/embedded_test_server/embedded_test_server.h"
#include "testing/perf/perf_test.h"
static const char kMainWebrtcTestHtmlPage[] =
"/webrtc/webrtc_jsep01_test.html";
// Performance browsertest for WebRTC. This test is manual since it takes long
// to execute and requires the reference files provided by the webrtc.DEPS
// solution (which is only available on WebRTC internal bots).
class WebRtcPerfBrowserTest : public WebRtcTestBase {
public:
void SetUpInProcessBrowserTestFixture() override {
DetectErrorsInJavaScript(); // Look for errors in our rather complex js.
}
void SetUpCommandLine(base::CommandLine* command_line) override {
// Ensure the infobar is enabled, since we expect that in this test.
EXPECT_FALSE(command_line->HasSwitch(switches::kUseFakeUIForMediaStream));
// Play a suitable, somewhat realistic video file.
base::FilePath input_video = test::GetReferenceFilesDir()
.Append(test::kReferenceFileName360p)
.AddExtension(test::kY4mFileExtension);
command_line->AppendSwitchPath(switches::kUseFileForFakeVideoCapture,
input_video);
command_line->AppendSwitch(switches::kUseFakeDeviceForMediaStream);
}
// Tries to extract data from peerConnectionDataStore in the webrtc-internals
// tab. The caller owns the parsed data. Returns NULL on failure.
base::DictionaryValue* GetWebrtcInternalsData(
content::WebContents* webrtc_internals_tab) {
std::string all_stats_json = ExecuteJavascript(
"window.domAutomationController.send("
" JSON.stringify(peerConnectionDataStore));",
webrtc_internals_tab);
scoped_ptr<base::Value> parsed_json =
base::JSONReader::Read(all_stats_json);
base::DictionaryValue* result;
if (parsed_json.get() && parsed_json->GetAsDictionary(&result)) {
ignore_result(parsed_json.release());
return result;
}
return NULL;
}
const base::DictionaryValue* GetDataOnPeerConnection(
const base::DictionaryValue* all_data,
int peer_connection_index) {
base::DictionaryValue::Iterator iterator(*all_data);
for (int i = 0; i < peer_connection_index && !iterator.IsAtEnd();
--peer_connection_index) {
iterator.Advance();
}
const base::DictionaryValue* result;
if (!iterator.IsAtEnd() && iterator.value().GetAsDictionary(&result))
return result;
return NULL;
}
scoped_ptr<base::DictionaryValue> MeasureWebRtcInternalsData(
int duration_msec) {
chrome::AddTabAt(browser(), GURL(), -1, true);
ui_test_utils::NavigateToURL(browser(), GURL("chrome://webrtc-internals"));
content::WebContents* webrtc_internals_tab =
browser()->tab_strip_model()->GetActiveWebContents();
test::SleepInJavascript(webrtc_internals_tab, duration_msec);
return scoped_ptr<base::DictionaryValue>(
GetWebrtcInternalsData(webrtc_internals_tab));
}
void RunsAudioVideoCall60SecsAndLogsInternalMetrics(
const std::string& video_codec) {
ASSERT_TRUE(test::HasReferenceFilesInCheckout());
ASSERT_TRUE(embedded_test_server()->Start());
ASSERT_GE(TestTimeouts::action_max_timeout().InSeconds(), 100)
<< "This is a long-running test; you must specify "
"--ui-test-action-max-timeout to have a value of at least 100000.";
content::WebContents* left_tab =
OpenTestPageAndGetUserMediaInNewTab(kMainWebrtcTestHtmlPage);
content::WebContents* right_tab =
OpenTestPageAndGetUserMediaInNewTab(kMainWebrtcTestHtmlPage);
SetupPeerconnectionWithLocalStream(left_tab);
SetupPeerconnectionWithLocalStream(right_tab);
NegotiateCall(left_tab, right_tab, video_codec);
StartDetectingVideo(left_tab, "remote-view");
StartDetectingVideo(right_tab, "remote-view");
WaitForVideoToPlay(left_tab);
WaitForVideoToPlay(right_tab);
// Let values stabilize, bandwidth ramp up, etc.
test::SleepInJavascript(left_tab, 60000);
// Start measurements.
scoped_ptr<base::DictionaryValue> all_data =
MeasureWebRtcInternalsData(10000);
ASSERT_TRUE(all_data.get() != NULL);
const base::DictionaryValue* first_pc_dict =
GetDataOnPeerConnection(all_data.get(), 0);
ASSERT_TRUE(first_pc_dict != NULL);
test::PrintBweForVideoMetrics(*first_pc_dict, "", video_codec);
test::PrintMetricsForAllStreams(*first_pc_dict, "", video_codec);
HangUp(left_tab);
HangUp(right_tab);
}
void RunsOneWayCall60SecsAndLogsInternalMetrics(
const std::string& video_codec) {
ASSERT_TRUE(test::HasReferenceFilesInCheckout());
ASSERT_TRUE(embedded_test_server()->Start());
ASSERT_GE(TestTimeouts::action_max_timeout().InSeconds(), 100)
<< "This is a long-running test; you must specify "
"--ui-test-action-max-timeout to have a value of at least 100000.";
content::WebContents* left_tab =
OpenTestPageAndGetUserMediaInNewTab(kMainWebrtcTestHtmlPage);
content::WebContents* right_tab =
OpenTestPageAndGetUserMediaInNewTab(kMainWebrtcTestHtmlPage);
SetupPeerconnectionWithLocalStream(left_tab);
SetupPeerconnectionWithoutLocalStream(right_tab);
NegotiateCall(left_tab, right_tab, video_codec);
// Remote video will only play in one tab since the call is one-way.
StartDetectingVideo(right_tab, "remote-view");
WaitForVideoToPlay(right_tab);
// Let values stabilize, bandwidth ramp up, etc.
test::SleepInJavascript(left_tab, 60000);
scoped_ptr<base::DictionaryValue> all_data =
MeasureWebRtcInternalsData(10000);
ASSERT_TRUE(all_data.get() != NULL);
// This assumes the sending peer connection is always listed first in the
// data store, and the receiving second.
const base::DictionaryValue* first_pc_dict =
GetDataOnPeerConnection(all_data.get(), 0);
ASSERT_TRUE(first_pc_dict != NULL);
test::PrintBweForVideoMetrics(*first_pc_dict, "_sendonly", video_codec);
test::PrintMetricsForAllStreams(*first_pc_dict, "_sendonly", video_codec);
const base::DictionaryValue* second_pc_dict =
GetDataOnPeerConnection(all_data.get(), 1);
ASSERT_TRUE(second_pc_dict != NULL);
test::PrintBweForVideoMetrics(*second_pc_dict, "_recvonly", video_codec);
test::PrintMetricsForAllStreams(*second_pc_dict, "_recvonly", video_codec);
HangUp(left_tab);
HangUp(right_tab);
}
};
// This is manual for its long execution time.
// The video codec name is now appended to result bucket (e.g. 'video_tx_VP8').
// TODO(asapersson): Keep test below using the default video codec (which do
// not have the codec name appended ('video_tx')) until new tests have been
// running for some time.
IN_PROC_BROWSER_TEST_F(
WebRtcPerfBrowserTest,
MANUAL_RunsAudioVideoCall60SecsAndLogsInternalMetricsDefault) {
RunsAudioVideoCall60SecsAndLogsInternalMetrics("");
}
IN_PROC_BROWSER_TEST_F(
WebRtcPerfBrowserTest,
MANUAL_RunsAudioVideoCall60SecsAndLogsInternalMetricsVp8) {
RunsAudioVideoCall60SecsAndLogsInternalMetrics("VP8");
}
IN_PROC_BROWSER_TEST_F(
WebRtcPerfBrowserTest,
MANUAL_RunsAudioVideoCall60SecsAndLogsInternalMetricsVp9) {
RunsAudioVideoCall60SecsAndLogsInternalMetrics("VP9");
}
#if BUILDFLAG(RTC_USE_H264)
IN_PROC_BROWSER_TEST_F(
WebRtcPerfBrowserTest,
MANUAL_RunsAudioVideoCall60SecsAndLogsInternalMetricsH264) {
// Only run test if run-time feature corresponding to |rtc_use_h264| is on.
if (!base::FeatureList::IsEnabled(content::kWebRtcH264WithOpenH264FFmpeg)) {
LOG(WARNING) << "Run-time feature WebRTC-H264WithOpenH264FFmpeg disabled. "
"Skipping WebRtcPerfBrowserTest.MANUAL_RunsAudioVideoCall60SecsAndLogs"
"InternalMetricsH264 (test \"OK\")";
return;
}
RunsAudioVideoCall60SecsAndLogsInternalMetrics("H264");
}
#endif // BUILDFLAG(RTC_USE_H264)
IN_PROC_BROWSER_TEST_F(
WebRtcPerfBrowserTest,
MANUAL_RunsOneWayCall60SecsAndLogsInternalMetricsDefault) {
RunsOneWayCall60SecsAndLogsInternalMetrics("");
}
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