blob: fe79f6df3c2c9c1ed104c04e090ff61f5cecef99 [file] [log] [blame]
/*
* 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<uint32_t> 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<uint8_t> 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<AbsoluteCaptureTime> 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