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}
210 lines
7.5 KiB
C++
210 lines
7.5 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 "testing/gmock/include/gmock/gmock.h"
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/base/scoped_ptr.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/video_engine/payload_router.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 PayloadRouterTest : public ::testing::Test {
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protected:
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virtual void SetUp() {
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payload_router_.reset(new PayloadRouter());
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}
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rtc::scoped_ptr<PayloadRouter> payload_router_;
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};
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TEST_F(PayloadRouterTest, SendOnOneModule) {
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MockRtpRtcp rtp;
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std::list<RtpRtcp*> modules(1, &rtp);
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payload_router_->SetSendingRtpModules(modules);
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uint8_t payload = 'a';
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FrameType frame_type = kVideoFrameKey;
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int8_t payload_type = 96;
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EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
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NULL))
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.Times(0);
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EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
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&payload, 1, NULL, NULL));
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payload_router_->set_active(true);
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EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
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NULL))
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.Times(1);
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EXPECT_TRUE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
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&payload, 1, NULL, NULL));
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payload_router_->set_active(false);
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EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
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NULL))
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.Times(0);
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EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
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&payload, 1, NULL, NULL));
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payload_router_->set_active(true);
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EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
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NULL))
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.Times(1);
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EXPECT_TRUE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
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&payload, 1, NULL, NULL));
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modules.clear();
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payload_router_->SetSendingRtpModules(modules);
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EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
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NULL))
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.Times(0);
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EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
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&payload, 1, NULL, NULL));
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}
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TEST_F(PayloadRouterTest, SendSimulcast) {
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MockRtpRtcp rtp_1;
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MockRtpRtcp rtp_2;
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std::list<RtpRtcp*> modules;
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modules.push_back(&rtp_1);
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modules.push_back(&rtp_2);
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payload_router_->SetSendingRtpModules(modules);
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uint8_t payload_1 = 'a';
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FrameType frame_type_1 = kVideoFrameKey;
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int8_t payload_type_1 = 96;
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RTPVideoHeader rtp_hdr_1;
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rtp_hdr_1.simulcastIdx = 0;
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payload_router_->set_active(true);
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EXPECT_CALL(rtp_1, SendOutgoingData(frame_type_1, payload_type_1, 0, 0, _, 1,
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NULL, &rtp_hdr_1))
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.Times(1);
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EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
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.Times(0);
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EXPECT_TRUE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0,
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&payload_1, 1, NULL, &rtp_hdr_1));
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uint8_t payload_2 = 'b';
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FrameType frame_type_2 = kVideoFrameDelta;
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int8_t payload_type_2 = 97;
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RTPVideoHeader rtp_hdr_2;
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rtp_hdr_2.simulcastIdx = 1;
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EXPECT_CALL(rtp_2, SendOutgoingData(frame_type_2, payload_type_2, 0, 0, _, 1,
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NULL, &rtp_hdr_2))
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.Times(1);
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EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
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.Times(0);
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EXPECT_TRUE(payload_router_->RoutePayload(frame_type_2, payload_type_2, 0, 0,
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&payload_2, 1, NULL, &rtp_hdr_2));
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// Inactive.
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payload_router_->set_active(false);
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EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
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.Times(0);
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EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
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.Times(0);
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EXPECT_FALSE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0,
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&payload_1, 1, NULL, &rtp_hdr_1));
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EXPECT_FALSE(payload_router_->RoutePayload(frame_type_2, payload_type_2, 0, 0,
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&payload_2, 1, NULL, &rtp_hdr_2));
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// Invalid simulcast index.
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payload_router_->set_active(true);
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EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
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.Times(0);
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EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
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.Times(0);
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rtp_hdr_1.simulcastIdx = 2;
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EXPECT_FALSE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0,
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&payload_1, 1, NULL, &rtp_hdr_1));
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}
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TEST_F(PayloadRouterTest, MaxPayloadLength) {
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// Without any limitations from the modules, verify we get the max payload
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// length for IP/UDP/SRTP with a MTU of 150 bytes.
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const size_t kDefaultMaxLength = 1500 - 20 - 8 - 12 - 4;
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EXPECT_EQ(kDefaultMaxLength, payload_router_->DefaultMaxPayloadLength());
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EXPECT_EQ(kDefaultMaxLength, payload_router_->MaxPayloadLength());
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MockRtpRtcp rtp_1;
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MockRtpRtcp rtp_2;
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std::list<RtpRtcp*> modules;
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modules.push_back(&rtp_1);
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modules.push_back(&rtp_2);
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payload_router_->SetSendingRtpModules(modules);
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// Modules return a higher length than the default value.
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EXPECT_CALL(rtp_1, MaxDataPayloadLength())
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.Times(1)
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.WillOnce(Return(kDefaultMaxLength + 10));
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EXPECT_CALL(rtp_2, MaxDataPayloadLength())
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.Times(1)
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.WillOnce(Return(kDefaultMaxLength + 10));
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EXPECT_EQ(kDefaultMaxLength, payload_router_->MaxPayloadLength());
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// The modules return a value lower than default.
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const size_t kTestMinPayloadLength = 1001;
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EXPECT_CALL(rtp_1, MaxDataPayloadLength())
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.Times(1)
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.WillOnce(Return(kTestMinPayloadLength + 10));
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EXPECT_CALL(rtp_2, MaxDataPayloadLength())
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.Times(1)
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.WillOnce(Return(kTestMinPayloadLength));
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EXPECT_EQ(kTestMinPayloadLength, payload_router_->MaxPayloadLength());
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}
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TEST_F(PayloadRouterTest, SetTargetSendBitrates) {
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MockRtpRtcp rtp_1;
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MockRtpRtcp rtp_2;
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std::list<RtpRtcp*> modules;
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modules.push_back(&rtp_1);
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modules.push_back(&rtp_2);
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payload_router_->SetSendingRtpModules(modules);
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const uint32_t bitrate_1 = 10000;
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const uint32_t bitrate_2 = 76543;
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std::vector<uint32_t> bitrates (2, bitrate_1);
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bitrates[1] = bitrate_2;
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EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
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.Times(1);
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EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
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.Times(1);
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payload_router_->SetTargetSendBitrates(bitrates);
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bitrates.resize(1);
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EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
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.Times(0);
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EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
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.Times(0);
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payload_router_->SetTargetSendBitrates(bitrates);
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bitrates.resize(3);
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bitrates[1] = bitrate_2;
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bitrates[2] = bitrate_1 + bitrate_2;
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EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
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.Times(1);
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EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
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.Times(1);
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payload_router_->SetTargetSendBitrates(bitrates);
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}
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} // namespace webrtc
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