/* * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ #include "api/rtp_packet_infos.h" #include "test/gmock.h" #include "test/gtest.h" namespace webrtc { TEST(RtpPacketInfoTest, Ssrc) { const uint32_t value = 4038189233; RtpPacketInfo lhs; RtpPacketInfo rhs; EXPECT_TRUE(lhs == rhs); EXPECT_FALSE(lhs != rhs); rhs.set_ssrc(value); EXPECT_EQ(rhs.ssrc(), value); EXPECT_FALSE(lhs == rhs); EXPECT_TRUE(lhs != rhs); lhs = rhs; EXPECT_TRUE(lhs == rhs); EXPECT_FALSE(lhs != rhs); rhs = RtpPacketInfo(); EXPECT_NE(rhs.ssrc(), value); rhs = RtpPacketInfo(value, {}, {}, {}, {}, {}); EXPECT_EQ(rhs.ssrc(), value); } TEST(RtpPacketInfoTest, Csrcs) { const std::vector value = {4038189233, 3016333617, 1207992985}; RtpPacketInfo lhs; RtpPacketInfo rhs; EXPECT_TRUE(lhs == rhs); EXPECT_FALSE(lhs != rhs); rhs.set_csrcs(value); EXPECT_EQ(rhs.csrcs(), value); EXPECT_FALSE(lhs == rhs); EXPECT_TRUE(lhs != rhs); lhs = rhs; EXPECT_TRUE(lhs == rhs); EXPECT_FALSE(lhs != rhs); rhs = RtpPacketInfo(); EXPECT_NE(rhs.csrcs(), value); rhs = RtpPacketInfo({}, value, {}, {}, {}, {}); EXPECT_EQ(rhs.csrcs(), value); } TEST(RtpPacketInfoTest, RtpTimestamp) { const uint32_t value = 4038189233; RtpPacketInfo lhs; RtpPacketInfo rhs; EXPECT_TRUE(lhs == rhs); EXPECT_FALSE(lhs != rhs); rhs.set_rtp_timestamp(value); EXPECT_EQ(rhs.rtp_timestamp(), value); EXPECT_FALSE(lhs == rhs); EXPECT_TRUE(lhs != rhs); lhs = rhs; EXPECT_TRUE(lhs == rhs); EXPECT_FALSE(lhs != rhs); rhs = RtpPacketInfo(); EXPECT_NE(rhs.rtp_timestamp(), value); rhs = RtpPacketInfo({}, {}, value, {}, {}, {}); EXPECT_EQ(rhs.rtp_timestamp(), value); } TEST(RtpPacketInfoTest, AudioLevel) { const absl::optional value = 31; RtpPacketInfo lhs; RtpPacketInfo rhs; EXPECT_TRUE(lhs == rhs); EXPECT_FALSE(lhs != rhs); rhs.set_audio_level(value); EXPECT_EQ(rhs.audio_level(), value); EXPECT_FALSE(lhs == rhs); EXPECT_TRUE(lhs != rhs); lhs = rhs; EXPECT_TRUE(lhs == rhs); EXPECT_FALSE(lhs != rhs); rhs = RtpPacketInfo(); EXPECT_NE(rhs.audio_level(), value); rhs = RtpPacketInfo({}, {}, {}, value, {}, {}); EXPECT_EQ(rhs.audio_level(), value); } TEST(RtpPacketInfoTest, AbsoluteCaptureTime) { const absl::optional value = AbsoluteCaptureTime{12, 34}; RtpPacketInfo lhs; RtpPacketInfo rhs; EXPECT_TRUE(lhs == rhs); EXPECT_FALSE(lhs != rhs); rhs.set_absolute_capture_time(value); EXPECT_EQ(rhs.absolute_capture_time(), value); EXPECT_FALSE(lhs == rhs); EXPECT_TRUE(lhs != rhs); lhs = rhs; EXPECT_TRUE(lhs == rhs); EXPECT_FALSE(lhs != rhs); rhs = RtpPacketInfo(); EXPECT_NE(rhs.absolute_capture_time(), value); rhs = RtpPacketInfo({}, {}, {}, {}, value, {}); EXPECT_EQ(rhs.absolute_capture_time(), value); } TEST(RtpPacketInfoTest, ReceiveTimeMs) { const int64_t value = 8868963877546349045LL; RtpPacketInfo lhs; RtpPacketInfo rhs; EXPECT_TRUE(lhs == rhs); EXPECT_FALSE(lhs != rhs); rhs.set_receive_time_ms(value); EXPECT_EQ(rhs.receive_time_ms(), value); EXPECT_FALSE(lhs == rhs); EXPECT_TRUE(lhs != rhs); lhs = rhs; EXPECT_TRUE(lhs == rhs); EXPECT_FALSE(lhs != rhs); rhs = RtpPacketInfo(); EXPECT_NE(rhs.receive_time_ms(), value); rhs = RtpPacketInfo({}, {}, {}, {}, {}, value); EXPECT_EQ(rhs.receive_time_ms(), value); } } // namespace webrtc