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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}
127 lines
3.4 KiB
C++
127 lines
3.4 KiB
C++
/*
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* Copyright (c) 2013 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 "test/rtp_file_reader.h"
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#include <map>
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#include <memory>
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#include "modules/rtp_rtcp/source/rtp_utility.h"
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#include "test/gtest.h"
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#include "test/testsupport/file_utils.h"
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namespace webrtc {
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class TestRtpFileReader : public ::testing::Test {
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public:
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void Init(const std::string& filename, bool headers_only_file) {
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std::string filepath =
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test::ResourcePath("video_coding/" + filename, "rtp");
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rtp_packet_source_.reset(
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test::RtpFileReader::Create(test::RtpFileReader::kRtpDump, filepath));
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ASSERT_TRUE(rtp_packet_source_.get() != NULL);
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headers_only_file_ = headers_only_file;
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}
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int CountRtpPackets() {
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test::RtpPacket packet;
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int c = 0;
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while (rtp_packet_source_->NextPacket(&packet)) {
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if (headers_only_file_)
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EXPECT_LT(packet.length, packet.original_length);
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else
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EXPECT_EQ(packet.length, packet.original_length);
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c++;
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}
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return c;
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}
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private:
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std::unique_ptr<test::RtpFileReader> rtp_packet_source_;
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bool headers_only_file_;
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};
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TEST_F(TestRtpFileReader, Test60Packets) {
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Init("pltype103", false);
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EXPECT_EQ(60, CountRtpPackets());
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}
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TEST_F(TestRtpFileReader, Test60PacketsHeaderOnly) {
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Init("pltype103_header_only", true);
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EXPECT_EQ(60, CountRtpPackets());
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}
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typedef std::map<uint32_t, int> PacketsPerSsrc;
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class TestPcapFileReader : public ::testing::Test {
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public:
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void Init(const std::string& filename) {
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std::string filepath =
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test::ResourcePath("video_coding/" + filename, "pcap");
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rtp_packet_source_.reset(
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test::RtpFileReader::Create(test::RtpFileReader::kPcap, filepath));
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ASSERT_TRUE(rtp_packet_source_.get() != NULL);
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}
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int CountRtpPackets() {
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int c = 0;
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test::RtpPacket packet;
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while (rtp_packet_source_->NextPacket(&packet)) {
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EXPECT_EQ(packet.length, packet.original_length);
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c++;
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}
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return c;
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}
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PacketsPerSsrc CountRtpPacketsPerSsrc() {
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PacketsPerSsrc pps;
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test::RtpPacket packet;
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while (rtp_packet_source_->NextPacket(&packet)) {
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RtpUtility::RtpHeaderParser rtp_header_parser(packet.data, packet.length);
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webrtc::RTPHeader header;
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if (!rtp_header_parser.RTCP() &&
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rtp_header_parser.Parse(&header, nullptr)) {
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pps[header.ssrc]++;
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}
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}
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return pps;
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}
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private:
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std::unique_ptr<test::RtpFileReader> rtp_packet_source_;
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};
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TEST_F(TestPcapFileReader, TestEthernetIIFrame) {
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Init("frame-ethernet-ii");
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EXPECT_EQ(368, CountRtpPackets());
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}
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TEST_F(TestPcapFileReader, TestLoopbackFrame) {
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Init("frame-loopback");
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EXPECT_EQ(491, CountRtpPackets());
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}
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TEST_F(TestPcapFileReader, TestTwoSsrc) {
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Init("ssrcs-2");
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PacketsPerSsrc pps = CountRtpPacketsPerSsrc();
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EXPECT_EQ(2UL, pps.size());
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EXPECT_EQ(370, pps[0x78d48f61]);
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EXPECT_EQ(60, pps[0xae94130b]);
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}
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TEST_F(TestPcapFileReader, TestThreeSsrc) {
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Init("ssrcs-3");
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PacketsPerSsrc pps = CountRtpPacketsPerSsrc();
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EXPECT_EQ(3UL, pps.size());
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EXPECT_EQ(162, pps[0x938c5eaa]);
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EXPECT_EQ(113, pps[0x59fe6ef0]);
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EXPECT_EQ(61, pps[0xed2bd2ac]);
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}
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} // namespace webrtc
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