webrtc_video_high_bitrate_browsertest.cc

// Copyright 2018 The Chromium Authors
// 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 "content/public/test/browser_test.h"
#include "media/base/media_switches.h"
#include "testing/perf/perf_test.h"
#include "ui/gl/gl_switches.h"

namespace {

static const char kMainWebrtcTestHtmlPage[] = …;
static const char kInboundRtp[] = …;
static const char kOutboundRtp[] = …;

// Sums up "RTC[In/Out]boundRTPStreamStats.bytes_[received/sent]" values.
double GetTotalRTPStreamBytes(content::TestStatsReportDictionary* report,
                              const char* type,
                              const char* media_type) { … }

double GetVideoBytesSent(content::TestStatsReportDictionary* report) { … }

double GetVideoBytesReceived(content::TestStatsReportDictionary* report) { … }

}  // 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 { … };

IN_PROC_BROWSER_TEST_F(WebRtcVideoHighBitrateBrowserTest,
                       MANUAL_HighBitrateEncodeDecode) { … }

}  // namespace content
chromium/chrome/browser/media/webrtc/webrtc_video_high_bitrate_browsertest.cc
