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a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
152 lines
3.1 KiB
C++
152 lines
3.1 KiB
C++
/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "api/rtp_packet_infos.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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namespace webrtc {
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TEST(RtpPacketInfoTest, Ssrc) {
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const uint32_t value = 4038189233;
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RtpPacketInfo lhs;
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RtpPacketInfo rhs;
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EXPECT_TRUE(lhs == rhs);
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EXPECT_FALSE(lhs != rhs);
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rhs.set_ssrc(value);
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EXPECT_EQ(rhs.ssrc(), value);
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EXPECT_FALSE(lhs == rhs);
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EXPECT_TRUE(lhs != rhs);
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lhs = rhs;
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EXPECT_TRUE(lhs == rhs);
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EXPECT_FALSE(lhs != rhs);
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rhs = RtpPacketInfo();
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EXPECT_NE(rhs.ssrc(), value);
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rhs = RtpPacketInfo(value, {}, {}, {}, {});
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EXPECT_EQ(rhs.ssrc(), value);
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}
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TEST(RtpPacketInfoTest, Csrcs) {
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const std::vector<uint32_t> value = {4038189233, 3016333617, 1207992985};
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RtpPacketInfo lhs;
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RtpPacketInfo rhs;
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EXPECT_TRUE(lhs == rhs);
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EXPECT_FALSE(lhs != rhs);
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rhs.set_csrcs(value);
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EXPECT_EQ(rhs.csrcs(), value);
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EXPECT_FALSE(lhs == rhs);
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EXPECT_TRUE(lhs != rhs);
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lhs = rhs;
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EXPECT_TRUE(lhs == rhs);
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EXPECT_FALSE(lhs != rhs);
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rhs = RtpPacketInfo();
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EXPECT_NE(rhs.csrcs(), value);
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rhs = RtpPacketInfo({}, value, {}, {}, {});
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EXPECT_EQ(rhs.csrcs(), value);
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}
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TEST(RtpPacketInfoTest, RtpTimestamp) {
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const uint32_t value = 4038189233;
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RtpPacketInfo lhs;
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RtpPacketInfo rhs;
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EXPECT_TRUE(lhs == rhs);
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EXPECT_FALSE(lhs != rhs);
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rhs.set_rtp_timestamp(value);
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EXPECT_EQ(rhs.rtp_timestamp(), value);
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EXPECT_FALSE(lhs == rhs);
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EXPECT_TRUE(lhs != rhs);
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lhs = rhs;
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EXPECT_TRUE(lhs == rhs);
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EXPECT_FALSE(lhs != rhs);
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rhs = RtpPacketInfo();
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EXPECT_NE(rhs.rtp_timestamp(), value);
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rhs = RtpPacketInfo({}, {}, value, {}, {});
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EXPECT_EQ(rhs.rtp_timestamp(), value);
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}
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TEST(RtpPacketInfoTest, AudioLevel) {
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const absl::optional<uint8_t> value = 31;
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RtpPacketInfo lhs;
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RtpPacketInfo rhs;
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EXPECT_TRUE(lhs == rhs);
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EXPECT_FALSE(lhs != rhs);
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rhs.set_audio_level(value);
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EXPECT_EQ(rhs.audio_level(), value);
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EXPECT_FALSE(lhs == rhs);
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EXPECT_TRUE(lhs != rhs);
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lhs = rhs;
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EXPECT_TRUE(lhs == rhs);
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EXPECT_FALSE(lhs != rhs);
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rhs = RtpPacketInfo();
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EXPECT_NE(rhs.audio_level(), value);
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rhs = RtpPacketInfo({}, {}, {}, value, {});
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EXPECT_EQ(rhs.audio_level(), value);
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}
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TEST(RtpPacketInfoTest, ReceiveTimeMs) {
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const int64_t value = 8868963877546349045LL;
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RtpPacketInfo lhs;
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RtpPacketInfo rhs;
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EXPECT_TRUE(lhs == rhs);
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EXPECT_FALSE(lhs != rhs);
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rhs.set_receive_time_ms(value);
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EXPECT_EQ(rhs.receive_time_ms(), value);
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EXPECT_FALSE(lhs == rhs);
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EXPECT_TRUE(lhs != rhs);
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lhs = rhs;
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EXPECT_TRUE(lhs == rhs);
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EXPECT_FALSE(lhs != rhs);
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rhs = RtpPacketInfo();
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EXPECT_NE(rhs.receive_time_ms(), value);
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rhs = RtpPacketInfo({}, {}, {}, {}, value);
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EXPECT_EQ(rhs.receive_time_ms(), value);
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}
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} // namespace webrtc
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