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platform-external-webrtc/webrtc/video_engine/vie_remb_unittest.cc
Henrik Kjellander ff761fba82 modules: more interface -> include renames 
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// This file includes unit tests for ViERemb.
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include <vector>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
#include "webrtc/modules/utility/include/mock/mock_process_thread.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/video_engine/vie_remb.h"
using ::testing::_;
using ::testing::AnyNumber;
using ::testing::NiceMock;
using ::testing::Return;
namespace webrtc {
class ViERembTest : public ::testing::Test {
protected:
virtual void SetUp() {
TickTime::UseFakeClock(12345);
process_thread_.reset(new NiceMock<MockProcessThread>);
vie_remb_.reset(new VieRemb());
}
rtc::scoped_ptr<MockProcessThread> process_thread_;
rtc::scoped_ptr<VieRemb> vie_remb_;
};
TEST_F(ViERembTest, OneModuleTestForSendingRemb) {
MockRtpRtcp rtp;
vie_remb_->AddReceiveChannel(&rtp);
vie_remb_->AddRembSender(&rtp);
const unsigned int bitrate_estimate = 456;
unsigned int ssrc = 1234;
std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
TickTime::AdvanceFakeClock(1000);
EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, ssrcs))
.Times(1);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
// Lower bitrate to send another REMB packet.
EXPECT_CALL(rtp, SetREMBData(bitrate_estimate - 100, ssrcs))
.Times(1);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate - 100);
vie_remb_->RemoveReceiveChannel(&rtp);
vie_remb_->RemoveRembSender(&rtp);
}
TEST_F(ViERembTest, LowerEstimateToSendRemb) {
MockRtpRtcp rtp;
vie_remb_->AddReceiveChannel(&rtp);
vie_remb_->AddRembSender(&rtp);
unsigned int bitrate_estimate = 456;
unsigned int ssrc = 1234;
std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
// Call OnReceiveBitrateChanged twice to get a first estimate.
TickTime::AdvanceFakeClock(1000);
EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, ssrcs))
.Times(1);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
// Lower the estimate with more than 3% to trigger a call to SetREMBData right
// away.
bitrate_estimate = bitrate_estimate - 100;
EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, ssrcs))
.Times(1);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
}
TEST_F(ViERembTest, VerifyIncreasingAndDecreasing) {
MockRtpRtcp rtp_0;
MockRtpRtcp rtp_1;
vie_remb_->AddReceiveChannel(&rtp_0);
vie_remb_->AddRembSender(&rtp_0);
vie_remb_->AddReceiveChannel(&rtp_1);
unsigned int bitrate_estimate[] = { 456, 789 };
unsigned int ssrc[] = { 1234, 5678 };
std::vector<unsigned int> ssrcs(ssrc, ssrc + sizeof(ssrc) / sizeof(ssrc[0]));
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate[0]);
// Call OnReceiveBitrateChanged twice to get a first estimate.
EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate[0], ssrcs))
.Times(1);
TickTime::AdvanceFakeClock(1000);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate[0]);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate[1] + 100);
// Lower the estimate to trigger a callback.
EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate[1], ssrcs))
.Times(1);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate[1]);
vie_remb_->RemoveReceiveChannel(&rtp_0);
vie_remb_->RemoveRembSender(&rtp_0);
vie_remb_->RemoveReceiveChannel(&rtp_1);
}
TEST_F(ViERembTest, NoRembForIncreasedBitrate) {
MockRtpRtcp rtp_0;
MockRtpRtcp rtp_1;
vie_remb_->AddReceiveChannel(&rtp_0);
vie_remb_->AddRembSender(&rtp_0);
vie_remb_->AddReceiveChannel(&rtp_1);
unsigned int bitrate_estimate = 456;
unsigned int ssrc[] = { 1234, 5678 };
std::vector<unsigned int> ssrcs(ssrc, ssrc + sizeof(ssrc) / sizeof(ssrc[0]));
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
// Call OnReceiveBitrateChanged twice to get a first estimate.
TickTime::AdvanceFakeClock(1000);
EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate, ssrcs))
.Times(1);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
// Increased estimate shouldn't trigger a callback right away.
EXPECT_CALL(rtp_0, SetREMBData(_, _))
.Times(0);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate + 1);
// Decreasing the estimate less than 3% shouldn't trigger a new callback.
EXPECT_CALL(rtp_0, SetREMBData(_, _))
.Times(0);
int lower_estimate = bitrate_estimate * 98 / 100;
vie_remb_->OnReceiveBitrateChanged(ssrcs, lower_estimate);
vie_remb_->RemoveReceiveChannel(&rtp_1);
vie_remb_->RemoveReceiveChannel(&rtp_0);
vie_remb_->RemoveRembSender(&rtp_0);
}
TEST_F(ViERembTest, ChangeSendRtpModule) {
MockRtpRtcp rtp_0;
MockRtpRtcp rtp_1;
vie_remb_->AddReceiveChannel(&rtp_0);
vie_remb_->AddRembSender(&rtp_0);
vie_remb_->AddReceiveChannel(&rtp_1);
unsigned int bitrate_estimate = 456;
unsigned int ssrc[] = { 1234, 5678 };
std::vector<unsigned int> ssrcs(ssrc, ssrc + sizeof(ssrc) / sizeof(ssrc[0]));
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
// Call OnReceiveBitrateChanged twice to get a first estimate.
TickTime::AdvanceFakeClock(1000);
EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate, ssrcs))
.Times(1);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
// Decrease estimate to trigger a REMB.
bitrate_estimate = bitrate_estimate - 100;
EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate, ssrcs))
.Times(1);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
// Remove the sending module, add it again -> should get remb on the second
// module.
vie_remb_->RemoveRembSender(&rtp_0);
vie_remb_->AddRembSender(&rtp_1);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
bitrate_estimate = bitrate_estimate - 100;
EXPECT_CALL(rtp_1, SetREMBData(bitrate_estimate, ssrcs))
.Times(1);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
vie_remb_->RemoveReceiveChannel(&rtp_0);
vie_remb_->RemoveReceiveChannel(&rtp_1);
}
TEST_F(ViERembTest, OnlyOneRembForDoubleProcess) {
MockRtpRtcp rtp;
unsigned int bitrate_estimate = 456;
unsigned int ssrc = 1234;
std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1);
vie_remb_->AddReceiveChannel(&rtp);
vie_remb_->AddRembSender(&rtp);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
// Call OnReceiveBitrateChanged twice to get a first estimate.
TickTime::AdvanceFakeClock(1000);
EXPECT_CALL(rtp, SetREMBData(_, _))
.Times(1);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
// Lower the estimate, should trigger a call to SetREMBData right away.
bitrate_estimate = bitrate_estimate - 100;
EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, ssrcs))
.Times(1);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
// Call OnReceiveBitrateChanged again, this should not trigger a new callback.
EXPECT_CALL(rtp, SetREMBData(_, _))
.Times(0);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
vie_remb_->RemoveReceiveChannel(&rtp);
vie_remb_->RemoveRembSender(&rtp);
}
// Only register receiving modules and make sure we fallback to trigger a REMB
// packet on this one.
TEST_F(ViERembTest, NoSendingRtpModule) {
MockRtpRtcp rtp;
vie_remb_->AddReceiveChannel(&rtp);
unsigned int bitrate_estimate = 456;
unsigned int ssrc = 1234;
std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
// Call OnReceiveBitrateChanged twice to get a first estimate.
TickTime::AdvanceFakeClock(1000);
EXPECT_CALL(rtp, SetREMBData(_, _))
.Times(1);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
// Lower the estimate to trigger a new packet REMB packet.
bitrate_estimate = bitrate_estimate - 100;
EXPECT_CALL(rtp, SetREMBData(_, _))
.Times(1);
vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
}
} // namespace webrtc
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