blob: 2bc096dbaab40d6eee42f11c5e43456377d1af00 [file] [log] [blame]
mflodman@webrtc.org02270cd2015-02-06 13:10:19 +00001/*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
mflodman@webrtc.org02270cd2015-02-06 13:10:19 +000011#include "testing/gmock/include/gmock/gmock.h"
12#include "testing/gtest/include/gtest/gtest.h"
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +000013#include "webrtc/base/scoped_ptr.h"
Henrik Kjellanderff761fb2015-11-04 08:31:52 +010014#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
mflodman@webrtc.org02270cd2015-02-06 13:10:19 +000015#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
Peter Boström7623ce42015-12-09 12:13:30 +010016#include "webrtc/video/payload_router.h"
mflodman@webrtc.org02270cd2015-02-06 13:10:19 +000017
18using ::testing::_;
19using ::testing::AnyNumber;
20using ::testing::NiceMock;
21using ::testing::Return;
22
23namespace webrtc {
24
25class PayloadRouterTest : public ::testing::Test {
26 protected:
27 virtual void SetUp() {
28 payload_router_.reset(new PayloadRouter());
29 }
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +000030 rtc::scoped_ptr<PayloadRouter> payload_router_;
mflodman@webrtc.org02270cd2015-02-06 13:10:19 +000031};
32
33TEST_F(PayloadRouterTest, SendOnOneModule) {
34 MockRtpRtcp rtp;
Peter Boström404686a2016-02-11 23:37:26 +010035 std::vector<RtpRtcp*> modules(1, &rtp);
mflodman@webrtc.org02270cd2015-02-06 13:10:19 +000036
37 payload_router_->SetSendingRtpModules(modules);
38
39 uint8_t payload = 'a';
40 FrameType frame_type = kVideoFrameKey;
41 int8_t payload_type = 96;
42
43 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
44 NULL))
45 .Times(0);
46 EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
47 &payload, 1, NULL, NULL));
48
49 payload_router_->set_active(true);
50 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
51 NULL))
52 .Times(1);
53 EXPECT_TRUE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
54 &payload, 1, NULL, NULL));
55
56 payload_router_->set_active(false);
57 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
58 NULL))
59 .Times(0);
60 EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
61 &payload, 1, NULL, NULL));
62
63 payload_router_->set_active(true);
64 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
65 NULL))
66 .Times(1);
67 EXPECT_TRUE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
68 &payload, 1, NULL, NULL));
69
70 modules.clear();
71 payload_router_->SetSendingRtpModules(modules);
72 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
73 NULL))
74 .Times(0);
75 EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
76 &payload, 1, NULL, NULL));
77}
78
79TEST_F(PayloadRouterTest, SendSimulcast) {
80 MockRtpRtcp rtp_1;
81 MockRtpRtcp rtp_2;
Peter Boström404686a2016-02-11 23:37:26 +010082 std::vector<RtpRtcp*> modules;
mflodman@webrtc.org02270cd2015-02-06 13:10:19 +000083 modules.push_back(&rtp_1);
84 modules.push_back(&rtp_2);
85
86 payload_router_->SetSendingRtpModules(modules);
87
88 uint8_t payload_1 = 'a';
89 FrameType frame_type_1 = kVideoFrameKey;
90 int8_t payload_type_1 = 96;
91 RTPVideoHeader rtp_hdr_1;
92 rtp_hdr_1.simulcastIdx = 0;
93
94 payload_router_->set_active(true);
95 EXPECT_CALL(rtp_1, SendOutgoingData(frame_type_1, payload_type_1, 0, 0, _, 1,
96 NULL, &rtp_hdr_1))
97 .Times(1);
98 EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
99 .Times(0);
100 EXPECT_TRUE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0,
101 &payload_1, 1, NULL, &rtp_hdr_1));
102
103 uint8_t payload_2 = 'b';
104 FrameType frame_type_2 = kVideoFrameDelta;
105 int8_t payload_type_2 = 97;
106 RTPVideoHeader rtp_hdr_2;
107 rtp_hdr_2.simulcastIdx = 1;
108 EXPECT_CALL(rtp_2, SendOutgoingData(frame_type_2, payload_type_2, 0, 0, _, 1,
109 NULL, &rtp_hdr_2))
110 .Times(1);
111 EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
112 .Times(0);
113 EXPECT_TRUE(payload_router_->RoutePayload(frame_type_2, payload_type_2, 0, 0,
114 &payload_2, 1, NULL, &rtp_hdr_2));
115
mflodman@webrtc.org50e28162015-02-23 07:45:11 +0000116 // Inactive.
mflodman@webrtc.org02270cd2015-02-06 13:10:19 +0000117 payload_router_->set_active(false);
118 EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
119 .Times(0);
120 EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
121 .Times(0);
122 EXPECT_FALSE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0,
123 &payload_1, 1, NULL, &rtp_hdr_1));
124 EXPECT_FALSE(payload_router_->RoutePayload(frame_type_2, payload_type_2, 0, 0,
125 &payload_2, 1, NULL, &rtp_hdr_2));
mflodman@webrtc.org50e28162015-02-23 07:45:11 +0000126
127 // Invalid simulcast index.
128 payload_router_->set_active(true);
129 EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
130 .Times(0);
131 EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
132 .Times(0);
133 rtp_hdr_1.simulcastIdx = 2;
134 EXPECT_FALSE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0,
135 &payload_1, 1, NULL, &rtp_hdr_1));
mflodman@webrtc.org02270cd2015-02-06 13:10:19 +0000136}
137
mflodman@webrtc.orga4ef2ce2015-02-12 09:54:18 +0000138TEST_F(PayloadRouterTest, MaxPayloadLength) {
139 // Without any limitations from the modules, verify we get the max payload
140 // length for IP/UDP/SRTP with a MTU of 150 bytes.
141 const size_t kDefaultMaxLength = 1500 - 20 - 8 - 12 - 4;
142 EXPECT_EQ(kDefaultMaxLength, payload_router_->DefaultMaxPayloadLength());
143 EXPECT_EQ(kDefaultMaxLength, payload_router_->MaxPayloadLength());
144
145 MockRtpRtcp rtp_1;
146 MockRtpRtcp rtp_2;
Peter Boström404686a2016-02-11 23:37:26 +0100147 std::vector<RtpRtcp*> modules;
mflodman@webrtc.orga4ef2ce2015-02-12 09:54:18 +0000148 modules.push_back(&rtp_1);
149 modules.push_back(&rtp_2);
150 payload_router_->SetSendingRtpModules(modules);
151
152 // Modules return a higher length than the default value.
153 EXPECT_CALL(rtp_1, MaxDataPayloadLength())
154 .Times(1)
155 .WillOnce(Return(kDefaultMaxLength + 10));
156 EXPECT_CALL(rtp_2, MaxDataPayloadLength())
157 .Times(1)
158 .WillOnce(Return(kDefaultMaxLength + 10));
159 EXPECT_EQ(kDefaultMaxLength, payload_router_->MaxPayloadLength());
160
161 // The modules return a value lower than default.
162 const size_t kTestMinPayloadLength = 1001;
163 EXPECT_CALL(rtp_1, MaxDataPayloadLength())
164 .Times(1)
165 .WillOnce(Return(kTestMinPayloadLength + 10));
166 EXPECT_CALL(rtp_2, MaxDataPayloadLength())
167 .Times(1)
168 .WillOnce(Return(kTestMinPayloadLength));
169 EXPECT_EQ(kTestMinPayloadLength, payload_router_->MaxPayloadLength());
170}
mflodman@webrtc.org02270cd2015-02-06 13:10:19 +0000171
mflodman@webrtc.org50e28162015-02-23 07:45:11 +0000172TEST_F(PayloadRouterTest, SetTargetSendBitrates) {
173 MockRtpRtcp rtp_1;
174 MockRtpRtcp rtp_2;
Peter Boström404686a2016-02-11 23:37:26 +0100175 std::vector<RtpRtcp*> modules;
mflodman@webrtc.org50e28162015-02-23 07:45:11 +0000176 modules.push_back(&rtp_1);
177 modules.push_back(&rtp_2);
178 payload_router_->SetSendingRtpModules(modules);
179
180 const uint32_t bitrate_1 = 10000;
181 const uint32_t bitrate_2 = 76543;
kjellander@webrtc.org0fcaf992015-11-26 15:24:52 +0100182 std::vector<uint32_t> bitrates(2, bitrate_1);
mflodman@webrtc.org50e28162015-02-23 07:45:11 +0000183 bitrates[1] = bitrate_2;
184 EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
185 .Times(1);
186 EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
187 .Times(1);
188 payload_router_->SetTargetSendBitrates(bitrates);
189
190 bitrates.resize(1);
191 EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
192 .Times(0);
193 EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
194 .Times(0);
195 payload_router_->SetTargetSendBitrates(bitrates);
196
197 bitrates.resize(3);
198 bitrates[1] = bitrate_2;
199 bitrates[2] = bitrate_1 + bitrate_2;
200 EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
201 .Times(1);
202 EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
203 .Times(1);
204 payload_router_->SetTargetSendBitrates(bitrates);
Stefan Holmere5904162015-03-26 11:11:06 +0100205}
mflodman@webrtc.org02270cd2015-02-06 13:10:19 +0000206} // namespace webrtc