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1 // Copyright 2013 The Chromium Authors. All rights reserved. | |
2 // Use of this source code is governed by a BSD-style license that can be | |
3 // found in the LICENSE file. | |
4 | |
5 #include "media/cast/rtp_sender/rtp_packetizer/rtp_packetizer.h" | |
6 | |
7 #include "base/memory/scoped_ptr.h" | |
8 #include "base/test/simple_test_tick_clock.h" | |
9 #include "media/cast/cast_config.h" | |
10 #include "media/cast/pacing/paced_sender.h" | |
11 #include "media/cast/rtp_common/rtp_defines.h" | |
12 #include "media/cast/rtp_sender/packet_storage/packet_storage.h" | |
13 #include "media/cast/rtp_sender/rtp_packetizer/test/rtp_header_parser.h" | |
14 #include "testing/gmock/include/gmock/gmock.h" | |
15 | |
16 namespace media { | |
17 namespace cast { | |
18 | |
19 static const int kPayload = 127; | |
20 static const uint32 kTimestampMs = 10; | |
21 static const uint16 kSeqNum = 33; | |
22 static const int kMaxPacketLength = 1500; | |
23 static const int kSsrc = 0x12345; | |
24 static const unsigned int kFrameSize = 5000; | |
25 static const int kMaxPacketStorageTimeMs = 300; | |
26 | |
27 class TestRtpPacketTransport : public PacedPacketSender { | |
28 public: | |
29 explicit TestRtpPacketTransport(RtpPacketizerConfig config) | |
30 : config_(config), | |
31 sequence_number_(kSeqNum), | |
32 packets_sent_(0), | |
33 expected_number_of_packets_(0), | |
34 expected_packet_id_(0), | |
35 expected_frame_id_(0) {} | |
36 | |
37 void VerifyRtpHeader(const RtpCastHeader& rtp_header) { | |
38 VerifyCommonRtpHeader(rtp_header); | |
39 VerifyCastRtpHeader(rtp_header); | |
40 } | |
41 | |
42 void VerifyCommonRtpHeader(const RtpCastHeader& rtp_header) { | |
43 EXPECT_EQ(expected_number_of_packets_ == packets_sent_, | |
44 rtp_header.webrtc.header.markerBit); | |
45 EXPECT_EQ(kPayload, rtp_header.webrtc.header.payloadType); | |
46 EXPECT_EQ(sequence_number_, rtp_header.webrtc.header.sequenceNumber); | |
47 EXPECT_EQ(kTimestampMs * 90, rtp_header.webrtc.header.timestamp); | |
48 EXPECT_EQ(config_.ssrc, rtp_header.webrtc.header.ssrc); | |
49 EXPECT_EQ(0, rtp_header.webrtc.header.numCSRCs); | |
50 } | |
51 | |
52 void VerifyCastRtpHeader(const RtpCastHeader& rtp_header) { | |
53 EXPECT_FALSE(rtp_header.is_key_frame); | |
54 EXPECT_EQ(expected_frame_id_, rtp_header.frame_id); | |
55 EXPECT_EQ(expected_packet_id_, rtp_header.packet_id); | |
56 EXPECT_EQ(expected_number_of_packets_ - 1, rtp_header.max_packet_id); | |
57 EXPECT_TRUE(rtp_header.is_reference); | |
58 EXPECT_EQ(expected_frame_id_ - 1u, rtp_header.reference_frame_id); | |
59 } | |
60 | |
61 virtual bool SendPackets(const PacketList& packets) OVERRIDE { | |
62 EXPECT_EQ(expected_number_of_packets_, static_cast<int>(packets.size())); | |
63 PacketList::const_iterator it = packets.begin(); | |
64 for (; it != packets.end(); ++it) { | |
65 ++packets_sent_; | |
66 RtpHeaderParser parser(it->data(), it->size()); | |
67 RtpCastHeader rtp_header; | |
68 parser.Parse(&rtp_header); | |
69 VerifyRtpHeader(rtp_header); | |
70 ++sequence_number_; | |
71 ++expected_packet_id_; | |
72 } | |
73 return true; | |
74 } | |
75 | |
76 virtual bool ResendPackets(const PacketList& packets) OVERRIDE { | |
77 EXPECT_TRUE(false); | |
78 return false; | |
79 } | |
80 | |
81 virtual bool SendRtcpPacket(const std::vector<uint8>& packet) OVERRIDE { | |
82 EXPECT_TRUE(false); | |
83 return false; | |
84 } | |
85 | |
86 void SetExpectedNumberOfPackets(int num) { | |
87 expected_number_of_packets_ = num; | |
88 } | |
89 | |
90 RtpPacketizerConfig config_; | |
91 uint32 sequence_number_; | |
92 int packets_sent_; | |
93 int expected_number_of_packets_; | |
94 // Assuming packets arrive in sequence. | |
95 int expected_packet_id_; | |
96 uint32 expected_frame_id_; | |
97 }; | |
98 | |
99 class RtpPacketizerTest : public ::testing::Test { | |
100 protected: | |
101 RtpPacketizerTest() | |
102 :video_frame_(), | |
103 packet_storage_(&testing_clock_, kMaxPacketStorageTimeMs) { | |
104 config_.sequence_number = kSeqNum; | |
105 config_.ssrc = kSsrc; | |
106 config_.payload_type = kPayload; | |
107 config_.max_payload_length = kMaxPacketLength; | |
108 transport_.reset(new TestRtpPacketTransport(config_)); | |
109 rtp_packetizer_.reset( | |
110 new RtpPacketizer(transport_.get(), &packet_storage_, config_)); | |
111 } | |
112 | |
113 virtual ~RtpPacketizerTest() {} | |
114 | |
115 virtual void SetUp() { | |
116 video_frame_.key_frame = false; | |
117 video_frame_.frame_id = 0; | |
118 video_frame_.last_referenced_frame_id = kStartFrameId; | |
119 video_frame_.data.assign(kFrameSize, 123); | |
120 } | |
121 | |
122 base::SimpleTestTickClock testing_clock_; | |
123 scoped_ptr<RtpPacketizer> rtp_packetizer_; | |
124 RtpPacketizerConfig config_; | |
125 scoped_ptr<TestRtpPacketTransport> transport_; | |
126 EncodedVideoFrame video_frame_; | |
127 PacketStorage packet_storage_; | |
128 }; | |
129 | |
130 TEST_F(RtpPacketizerTest, SendStandardPackets) { | |
131 int expected_num_of_packets = kFrameSize / kMaxPacketLength + 1; | |
132 transport_->SetExpectedNumberOfPackets(expected_num_of_packets); | |
133 | |
134 base::TimeTicks time; | |
135 time += base::TimeDelta::FromMilliseconds(kTimestampMs); | |
136 rtp_packetizer_->IncomingEncodedVideoFrame(&video_frame_, time); | |
137 } | |
138 | |
139 TEST_F(RtpPacketizerTest, Stats) { | |
140 EXPECT_FALSE(rtp_packetizer_->send_packets_count()); | |
141 EXPECT_FALSE(rtp_packetizer_->send_octet_count()); | |
142 // Insert packets at varying lengths. | |
143 int expected_num_of_packets = kFrameSize / kMaxPacketLength + 1; | |
144 transport_->SetExpectedNumberOfPackets(expected_num_of_packets); | |
145 | |
146 testing_clock_.Advance(base::TimeDelta::FromMilliseconds(kTimestampMs)); | |
147 rtp_packetizer_->IncomingEncodedVideoFrame(&video_frame_, | |
148 testing_clock_.NowTicks()); | |
149 EXPECT_EQ(expected_num_of_packets, rtp_packetizer_->send_packets_count()); | |
150 EXPECT_EQ(kFrameSize, rtp_packetizer_->send_octet_count()); | |
151 } | |
152 | |
153 } // namespace cast | |
154 } // namespace media | |
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