rtp_packet_infos_unittest.cc (3873B)
1 /* 2 * Copyright (c) 2019 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 11 #include "api/rtp_packet_infos.h" 12 13 #include <cstddef> 14 15 #include "api/rtp_headers.h" 16 #include "api/rtp_packet_info.h" 17 #include "api/units/timestamp.h" 18 #include "test/gmock.h" 19 #include "test/gtest.h" 20 21 namespace webrtc { 22 namespace { 23 24 using ::testing::ElementsAre; 25 using ::testing::SizeIs; 26 27 template <typename Iterator> 28 RtpPacketInfos::vector_type ToVector(Iterator begin, Iterator end) { 29 return RtpPacketInfos::vector_type(begin, end); 30 } 31 32 } // namespace 33 34 TEST(RtpPacketInfosTest, BasicFunctionality) { 35 RtpPacketInfo p0(/*ssrc=*/123, /*csrcs=*/{1, 2}, /*rtp_timestamp=*/89, 36 /*receive_time=*/Timestamp::Millis(7)); 37 p0.set_audio_level(5); 38 p0.set_absolute_capture_time(AbsoluteCaptureTime{ 39 .absolute_capture_timestamp = 45, .estimated_capture_clock_offset = 78}); 40 41 RtpPacketInfo p1(/*ssrc=*/456, /*csrcs=*/{3, 4}, /*rtp_timestamp=*/89, 42 /*receive_time=*/Timestamp::Millis(1)); 43 p1.set_audio_level(4); 44 p1.set_absolute_capture_time(AbsoluteCaptureTime{ 45 .absolute_capture_timestamp = 13, .estimated_capture_clock_offset = 21}); 46 47 RtpPacketInfo p2(/*ssrc=*/789, /*csrcs=*/{5, 6}, /*rtp_timestamp=*/88, 48 /*receive_time=*/Timestamp::Millis(7)); 49 p2.set_audio_level(1); 50 p2.set_absolute_capture_time(AbsoluteCaptureTime{ 51 .absolute_capture_timestamp = 99, .estimated_capture_clock_offset = 78}); 52 53 RtpPacketInfos x({p0, p1, p2}); 54 55 ASSERT_THAT(x, SizeIs(3)); 56 57 EXPECT_EQ(x[0], p0); 58 EXPECT_EQ(x[1], p1); 59 EXPECT_EQ(x[2], p2); 60 61 EXPECT_EQ(x.front(), p0); 62 EXPECT_EQ(x.back(), p2); 63 64 EXPECT_THAT(ToVector(x.begin(), x.end()), ElementsAre(p0, p1, p2)); 65 EXPECT_THAT(ToVector(x.rbegin(), x.rend()), ElementsAre(p2, p1, p0)); 66 67 EXPECT_THAT(ToVector(x.cbegin(), x.cend()), ElementsAre(p0, p1, p2)); 68 EXPECT_THAT(ToVector(x.crbegin(), x.crend()), ElementsAre(p2, p1, p0)); 69 70 EXPECT_FALSE(x.empty()); 71 } 72 73 TEST(RtpPacketInfosTest, CopyShareData) { 74 RtpPacketInfo p0(/*ssrc=*/123, /*csrcs=*/{1, 2}, /*rtp_timestamp=*/89, 75 /*receive_time=*/Timestamp::Millis(7)); 76 p0.set_audio_level(5); 77 p0.set_absolute_capture_time(AbsoluteCaptureTime{ 78 .absolute_capture_timestamp = 45, .estimated_capture_clock_offset = 78}); 79 80 RtpPacketInfo p1(/*ssrc=*/456, /*csrcs=*/{3, 4}, /*rtp_timestamp=*/89, 81 /*receive_time=*/Timestamp::Millis(1)); 82 p1.set_audio_level(4); 83 p1.set_absolute_capture_time(AbsoluteCaptureTime{ 84 .absolute_capture_timestamp = 13, .estimated_capture_clock_offset = 21}); 85 86 RtpPacketInfo p2(/*ssrc=*/789, /*csrcs=*/{5, 6}, /*rtp_timestamp=*/88, 87 /*receive_time=*/Timestamp::Millis(7)); 88 p2.set_audio_level(1); 89 p2.set_absolute_capture_time(AbsoluteCaptureTime{ 90 .absolute_capture_timestamp = 99, .estimated_capture_clock_offset = 78}); 91 92 RtpPacketInfos lhs({p0, p1, p2}); 93 RtpPacketInfos rhs = lhs; 94 95 ASSERT_THAT(lhs, SizeIs(3)); 96 ASSERT_THAT(rhs, SizeIs(3)); 97 98 for (size_t i = 0; i < lhs.size(); ++i) { 99 EXPECT_EQ(lhs[i], rhs[i]); 100 } 101 102 EXPECT_EQ(lhs.front(), rhs.front()); 103 EXPECT_EQ(lhs.back(), rhs.back()); 104 105 EXPECT_EQ(lhs.begin(), rhs.begin()); 106 EXPECT_EQ(lhs.end(), rhs.end()); 107 EXPECT_EQ(lhs.rbegin(), rhs.rbegin()); 108 EXPECT_EQ(lhs.rend(), rhs.rend()); 109 110 EXPECT_EQ(lhs.cbegin(), rhs.cbegin()); 111 EXPECT_EQ(lhs.cend(), rhs.cend()); 112 EXPECT_EQ(lhs.crbegin(), rhs.crbegin()); 113 EXPECT_EQ(lhs.crend(), rhs.crend()); 114 115 EXPECT_EQ(lhs.empty(), rhs.empty()); 116 } 117 118 } // namespace webrtc