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https://github.com/mollyim/webrtc.git
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d2a6686a10
This change adds classes so that we later can plumb information about received packets to each audio and video frame. It's not wired up to do anything yet. Bug: webrtc:10668 Change-Id: I962df493a76692f668314f78d6792d7636c5a31b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138203 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28138}
180 lines
3.7 KiB
C++
180 lines
3.7 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, SequenceNumber) {
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const uint16_t value = 20238;
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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_sequence_number(value);
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EXPECT_EQ(rhs.sequence_number(), 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.sequence_number(), value);
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rhs = RtpPacketInfo({}, {}, value, {}, {}, {});
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EXPECT_EQ(rhs.sequence_number(), 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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