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https://github.com/mollyim/webrtc.git
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a1d035655e
New ctor added without optional and media specific fields. Bug: webrtc:10739, b/246753278 Change-Id: I7e15849aced6ed0a7ada725ea171a15ea1e9bc5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275941 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38124}
113 lines
3.7 KiB
C++
113 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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namespace {
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using ::testing::ElementsAre;
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using ::testing::SizeIs;
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template <typename Iterator>
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RtpPacketInfos::vector_type ToVector(Iterator begin, Iterator end) {
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return RtpPacketInfos::vector_type(begin, end);
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}
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} // namespace
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TEST(RtpPacketInfosTest, BasicFunctionality) {
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RtpPacketInfo p0(/*ssrc=*/123, /*csrcs=*/{1, 2}, /*rtp_timestamp=*/89,
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/*receive_time=*/Timestamp::Millis(7));
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p0.set_audio_level(5);
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p0.set_absolute_capture_time(AbsoluteCaptureTime{
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.absolute_capture_timestamp = 45, .estimated_capture_clock_offset = 78});
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RtpPacketInfo p1(/*ssrc=*/456, /*csrcs=*/{3, 4}, /*rtp_timestamp=*/89,
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/*receive_time=*/Timestamp::Millis(1));
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p1.set_audio_level(4);
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p1.set_absolute_capture_time(AbsoluteCaptureTime{
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.absolute_capture_timestamp = 13, .estimated_capture_clock_offset = 21});
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RtpPacketInfo p2(/*ssrc=*/789, /*csrcs=*/{5, 6}, /*rtp_timestamp=*/88,
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/*receive_time=*/Timestamp::Millis(7));
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p2.set_audio_level(1);
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p2.set_absolute_capture_time(AbsoluteCaptureTime{
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.absolute_capture_timestamp = 99, .estimated_capture_clock_offset = 78});
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RtpPacketInfos x({p0, p1, p2});
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ASSERT_THAT(x, SizeIs(3));
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EXPECT_EQ(x[0], p0);
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EXPECT_EQ(x[1], p1);
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EXPECT_EQ(x[2], p2);
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EXPECT_EQ(x.front(), p0);
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EXPECT_EQ(x.back(), p2);
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EXPECT_THAT(ToVector(x.begin(), x.end()), ElementsAre(p0, p1, p2));
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EXPECT_THAT(ToVector(x.rbegin(), x.rend()), ElementsAre(p2, p1, p0));
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EXPECT_THAT(ToVector(x.cbegin(), x.cend()), ElementsAre(p0, p1, p2));
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EXPECT_THAT(ToVector(x.crbegin(), x.crend()), ElementsAre(p2, p1, p0));
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EXPECT_FALSE(x.empty());
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}
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TEST(RtpPacketInfosTest, CopyShareData) {
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RtpPacketInfo p0(/*ssrc=*/123, /*csrcs=*/{1, 2}, /*rtp_timestamp=*/89,
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/*receive_time=*/Timestamp::Millis(7));
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p0.set_audio_level(5);
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p0.set_absolute_capture_time(AbsoluteCaptureTime{
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.absolute_capture_timestamp = 45, .estimated_capture_clock_offset = 78});
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RtpPacketInfo p1(/*ssrc=*/456, /*csrcs=*/{3, 4}, /*rtp_timestamp=*/89,
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/*receive_time=*/Timestamp::Millis(1));
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p1.set_audio_level(4);
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p1.set_absolute_capture_time(AbsoluteCaptureTime{
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.absolute_capture_timestamp = 13, .estimated_capture_clock_offset = 21});
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RtpPacketInfo p2(/*ssrc=*/789, /*csrcs=*/{5, 6}, /*rtp_timestamp=*/88,
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/*receive_time=*/Timestamp::Millis(7));
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p2.set_audio_level(1);
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p2.set_absolute_capture_time(AbsoluteCaptureTime{
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.absolute_capture_timestamp = 99, .estimated_capture_clock_offset = 78});
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RtpPacketInfos lhs({p0, p1, p2});
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RtpPacketInfos rhs = lhs;
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ASSERT_THAT(lhs, SizeIs(3));
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ASSERT_THAT(rhs, SizeIs(3));
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for (size_t i = 0; i < lhs.size(); ++i) {
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EXPECT_EQ(lhs[i], rhs[i]);
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}
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EXPECT_EQ(lhs.front(), rhs.front());
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EXPECT_EQ(lhs.back(), rhs.back());
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EXPECT_EQ(lhs.begin(), rhs.begin());
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EXPECT_EQ(lhs.end(), rhs.end());
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EXPECT_EQ(lhs.rbegin(), rhs.rbegin());
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EXPECT_EQ(lhs.rend(), rhs.rend());
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EXPECT_EQ(lhs.cbegin(), rhs.cbegin());
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EXPECT_EQ(lhs.cend(), rhs.cend());
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EXPECT_EQ(lhs.crbegin(), rhs.crbegin());
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EXPECT_EQ(lhs.crend(), rhs.crend());
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EXPECT_EQ(lhs.empty(), rhs.empty());
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}
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} // namespace webrtc
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