ff761fba82
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}
173 lines
7.2 KiB
C++
173 lines
7.2 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <list>
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#include "webrtc/base/checks.h"
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#include "testing/gmock/include/gmock/gmock.h"
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/modules/pacing/include/packet_router.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
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#include "webrtc/base/scoped_ptr.h"
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using ::testing::_;
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using ::testing::AnyNumber;
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using ::testing::NiceMock;
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using ::testing::Return;
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namespace webrtc {
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class PacketRouterTest : public ::testing::Test {
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public:
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PacketRouterTest() : packet_router_(new PacketRouter()) {}
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protected:
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const rtc::scoped_ptr<PacketRouter> packet_router_;
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};
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TEST_F(PacketRouterTest, TimeToSendPacket) {
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MockRtpRtcp rtp_1;
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MockRtpRtcp rtp_2;
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packet_router_->AddRtpModule(&rtp_1);
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packet_router_->AddRtpModule(&rtp_2);
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const uint16_t kSsrc1 = 1234;
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uint16_t sequence_number = 17;
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uint64_t timestamp = 7890;
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bool retransmission = false;
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// Send on the first module by letting rtp_1 be sending with correct ssrc.
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EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true));
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EXPECT_CALL(rtp_1, SSRC()).Times(1).WillOnce(Return(kSsrc1));
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EXPECT_CALL(rtp_1, TimeToSendPacket(kSsrc1, sequence_number, timestamp,
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retransmission))
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.Times(1)
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.WillOnce(Return(true));
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EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0);
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EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number,
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timestamp, retransmission));
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// Send on the second module by letting rtp_2 be sending, but not rtp_1.
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++sequence_number;
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timestamp += 30;
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retransmission = true;
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const uint16_t kSsrc2 = 4567;
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EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(false));
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EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
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EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2));
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EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _, _)).Times(0);
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EXPECT_CALL(rtp_2, TimeToSendPacket(kSsrc2, sequence_number, timestamp,
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retransmission))
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.Times(1)
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.WillOnce(Return(true));
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EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc2, sequence_number,
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timestamp, retransmission));
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// No module is sending, hence no packet should be sent.
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EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(false));
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EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _, _)).Times(0);
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EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(false));
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EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0);
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EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number,
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timestamp, retransmission));
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// Add a packet with incorrect ssrc and test it's dropped in the router.
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EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true));
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EXPECT_CALL(rtp_1, SSRC()).Times(1).WillOnce(Return(kSsrc1));
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EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
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EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2));
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EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _, _)).Times(0);
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EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0);
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EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1 + kSsrc2, sequence_number,
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timestamp, retransmission));
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packet_router_->RemoveRtpModule(&rtp_1);
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// rtp_1 has been removed, try sending a packet on that ssrc and make sure
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// it is dropped as expected by not expecting any calls to rtp_1.
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EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
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EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2));
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EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0);
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EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number,
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timestamp, retransmission));
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packet_router_->RemoveRtpModule(&rtp_2);
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}
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TEST_F(PacketRouterTest, TimeToSendPadding) {
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const uint16_t kSsrc1 = 1234;
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const uint16_t kSsrc2 = 4567;
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MockRtpRtcp rtp_1;
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EXPECT_CALL(rtp_1, SSRC()).WillRepeatedly(Return(kSsrc1));
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MockRtpRtcp rtp_2;
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EXPECT_CALL(rtp_2, SSRC()).WillRepeatedly(Return(kSsrc2));
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packet_router_->AddRtpModule(&rtp_1);
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packet_router_->AddRtpModule(&rtp_2);
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// Default configuration, sending padding on all modules sending media,
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// ordered by SSRC.
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const size_t requested_padding_bytes = 1000;
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const size_t sent_padding_bytes = 890;
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EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true));
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EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes))
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.Times(1)
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.WillOnce(Return(sent_padding_bytes));
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EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
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EXPECT_CALL(rtp_2,
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TimeToSendPadding(requested_padding_bytes - sent_padding_bytes))
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.Times(1)
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.WillOnce(Return(requested_padding_bytes - sent_padding_bytes));
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EXPECT_EQ(requested_padding_bytes,
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packet_router_->TimeToSendPadding(requested_padding_bytes));
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// Let only the second module be sending and verify the padding request is
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// routed there.
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EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(false));
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EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes)).Times(0);
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EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
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EXPECT_CALL(rtp_2, TimeToSendPadding(_))
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.Times(1)
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.WillOnce(Return(sent_padding_bytes));
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EXPECT_EQ(sent_padding_bytes,
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packet_router_->TimeToSendPadding(requested_padding_bytes));
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// No sending module at all.
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EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(false));
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EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes)).Times(0);
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EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(false));
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EXPECT_CALL(rtp_2, TimeToSendPadding(_)).Times(0);
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EXPECT_EQ(0u, packet_router_->TimeToSendPadding(requested_padding_bytes));
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packet_router_->RemoveRtpModule(&rtp_1);
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// rtp_1 has been removed, try sending padding and make sure rtp_1 isn't asked
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// to send by not expecting any calls. Instead verify rtp_2 is called.
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EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
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EXPECT_CALL(rtp_2, TimeToSendPadding(requested_padding_bytes)).Times(1);
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EXPECT_EQ(0u, packet_router_->TimeToSendPadding(requested_padding_bytes));
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packet_router_->RemoveRtpModule(&rtp_2);
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}
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TEST_F(PacketRouterTest, AllocateSequenceNumbers) {
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const uint16_t kStartSeq = 0xFFF0;
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const size_t kNumPackets = 32;
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packet_router_->SetTransportWideSequenceNumber(kStartSeq - 1);
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for (size_t i = 0; i < kNumPackets; ++i) {
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uint16_t seq = packet_router_->AllocateSequenceNumber();
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uint32_t expected_unwrapped_seq = static_cast<uint32_t>(kStartSeq) + i;
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EXPECT_EQ(static_cast<uint16_t>(expected_unwrapped_seq & 0xFFFF), seq);
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}
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}
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} // namespace webrtc
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