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/*
* 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 {
namespace {
using ::testing::ElementsAre;
using ::testing::SizeIs;
template <typename Iterator>
RtpPacketInfos::vector_type ToVector(Iterator begin, Iterator end) {
return RtpPacketInfos::vector_type(begin, end);
}
} // namespace
TEST(RtpPacketInfosTest, BasicFunctionality) {
RtpPacketInfo p0(/*ssrc=*/123, /*csrcs=*/{1, 2}, /*rtp_timestamp=*/89,
/*receive_time=*/Timestamp::Millis(7));
p0.set_audio_level(5);
p0.set_absolute_capture_time(AbsoluteCaptureTime{
.absolute_capture_timestamp = 45, .estimated_capture_clock_offset = 78});
RtpPacketInfo p1(/*ssrc=*/456, /*csrcs=*/{3, 4}, /*rtp_timestamp=*/89,
/*receive_time=*/Timestamp::Millis(1));
p1.set_audio_level(4);
p1.set_absolute_capture_time(AbsoluteCaptureTime{
.absolute_capture_timestamp = 13, .estimated_capture_clock_offset = 21});
RtpPacketInfo p2(/*ssrc=*/789, /*csrcs=*/{5, 6}, /*rtp_timestamp=*/88,
/*receive_time=*/Timestamp::Millis(7));
p2.set_audio_level(1);
p2.set_absolute_capture_time(AbsoluteCaptureTime{
.absolute_capture_timestamp = 99, .estimated_capture_clock_offset = 78});
RtpPacketInfos x({p0, p1, p2});
ASSERT_THAT(x, SizeIs(3));
EXPECT_EQ(x[0], p0);
EXPECT_EQ(x[1], p1);
EXPECT_EQ(x[2], p2);
EXPECT_EQ(x.front(), p0);
EXPECT_EQ(x.back(), p2);
EXPECT_THAT(ToVector(x.begin(), x.end()), ElementsAre(p0, p1, p2));
EXPECT_THAT(ToVector(x.rbegin(), x.rend()), ElementsAre(p2, p1, p0));
EXPECT_THAT(ToVector(x.cbegin(), x.cend()), ElementsAre(p0, p1, p2));
EXPECT_THAT(ToVector(x.crbegin(), x.crend()), ElementsAre(p2, p1, p0));
EXPECT_FALSE(x.empty());
}
TEST(RtpPacketInfosTest, CopyShareData) {
RtpPacketInfo p0(/*ssrc=*/123, /*csrcs=*/{1, 2}, /*rtp_timestamp=*/89,
/*receive_time=*/Timestamp::Millis(7));
p0.set_audio_level(5);
p0.set_absolute_capture_time(AbsoluteCaptureTime{
.absolute_capture_timestamp = 45, .estimated_capture_clock_offset = 78});
RtpPacketInfo p1(/*ssrc=*/456, /*csrcs=*/{3, 4}, /*rtp_timestamp=*/89,
/*receive_time=*/Timestamp::Millis(1));
p1.set_audio_level(4);
p1.set_absolute_capture_time(AbsoluteCaptureTime{
.absolute_capture_timestamp = 13, .estimated_capture_clock_offset = 21});
RtpPacketInfo p2(/*ssrc=*/789, /*csrcs=*/{5, 6}, /*rtp_timestamp=*/88,
/*receive_time=*/Timestamp::Millis(7));
p2.set_audio_level(1);
p2.set_absolute_capture_time(AbsoluteCaptureTime{
.absolute_capture_timestamp = 99, .estimated_capture_clock_offset = 78});
RtpPacketInfos lhs({p0, p1, p2});
RtpPacketInfos rhs = lhs;
ASSERT_THAT(lhs, SizeIs(3));
ASSERT_THAT(rhs, SizeIs(3));
for (size_t i = 0; i < lhs.size(); ++i) {
EXPECT_EQ(lhs[i], rhs[i]);
}
EXPECT_EQ(lhs.front(), rhs.front());
EXPECT_EQ(lhs.back(), rhs.back());
EXPECT_EQ(lhs.begin(), rhs.begin());
EXPECT_EQ(lhs.end(), rhs.end());
EXPECT_EQ(lhs.rbegin(), rhs.rbegin());
EXPECT_EQ(lhs.rend(), rhs.rend());
EXPECT_EQ(lhs.cbegin(), rhs.cbegin());
EXPECT_EQ(lhs.cend(), rhs.cend());
EXPECT_EQ(lhs.crbegin(), rhs.crbegin());
EXPECT_EQ(lhs.crend(), rhs.crend());
EXPECT_EQ(lhs.empty(), rhs.empty());
}
} // namespace webrtc
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