mirror of
https://github.com/mollyim/webrtc.git
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b9b146c9fe
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'audio call video':
#!/bin/bash
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
118 lines
3.9 KiB
C++
118 lines
3.9 KiB
C++
/*
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* Copyright (c) 2017 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 <cstdio>
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#include "call/rtp_rtcp_demuxer_helper.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/pli.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
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#include "rtc_base/arraysize.h"
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#include "rtc_base/buffer.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace {
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constexpr uint32_t kSsrc = 8374;
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} // namespace
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TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_ByePacket) {
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webrtc::rtcp::Bye rtcp_packet;
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rtcp_packet.SetSenderSsrc(kSsrc);
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rtc::Buffer raw_packet = rtcp_packet.Build();
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absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
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EXPECT_EQ(ssrc, kSsrc);
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}
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TEST(RtpRtcpDemuxerHelperTest,
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ParseRtcpPacketSenderSsrc_ExtendedReportsPacket) {
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webrtc::rtcp::ExtendedReports rtcp_packet;
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rtcp_packet.SetSenderSsrc(kSsrc);
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rtc::Buffer raw_packet = rtcp_packet.Build();
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absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
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EXPECT_EQ(ssrc, kSsrc);
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}
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TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_PsfbPacket) {
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webrtc::rtcp::Pli rtcp_packet; // Psfb is abstract; use a subclass.
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rtcp_packet.SetSenderSsrc(kSsrc);
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rtc::Buffer raw_packet = rtcp_packet.Build();
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absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
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EXPECT_EQ(ssrc, kSsrc);
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}
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TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_ReceiverReportPacket) {
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webrtc::rtcp::ReceiverReport rtcp_packet;
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rtcp_packet.SetSenderSsrc(kSsrc);
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rtc::Buffer raw_packet = rtcp_packet.Build();
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absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
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EXPECT_EQ(ssrc, kSsrc);
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}
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TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_RtpfbPacket) {
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// Rtpfb is abstract; use a subclass.
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webrtc::rtcp::RapidResyncRequest rtcp_packet;
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rtcp_packet.SetSenderSsrc(kSsrc);
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rtc::Buffer raw_packet = rtcp_packet.Build();
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absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
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EXPECT_EQ(ssrc, kSsrc);
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}
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TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_SenderReportPacket) {
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webrtc::rtcp::SenderReport rtcp_packet;
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rtcp_packet.SetSenderSsrc(kSsrc);
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rtc::Buffer raw_packet = rtcp_packet.Build();
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absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
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EXPECT_EQ(ssrc, kSsrc);
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}
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TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_MalformedRtcpPacket) {
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uint8_t garbage[100];
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memset(&garbage[0], 0, arraysize(garbage));
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absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(garbage);
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EXPECT_FALSE(ssrc);
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}
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TEST(RtpRtcpDemuxerHelperTest,
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ParseRtcpPacketSenderSsrc_RtcpMessageWithoutSenderSsrc) {
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webrtc::rtcp::ExtendedJitterReport rtcp_packet; // Has no sender SSRC.
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rtc::Buffer raw_packet = rtcp_packet.Build();
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absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
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EXPECT_FALSE(ssrc);
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}
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TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_TruncatedRtcpMessage) {
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webrtc::rtcp::Bye rtcp_packet;
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rtcp_packet.SetSenderSsrc(kSsrc);
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rtc::Buffer raw_packet = rtcp_packet.Build();
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constexpr size_t rtcp_length_bytes = 8;
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ASSERT_EQ(rtcp_length_bytes, raw_packet.size());
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absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(
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rtc::ArrayView<const uint8_t>(raw_packet.data(), rtcp_length_bytes - 1));
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EXPECT_FALSE(ssrc);
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}
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} // namespace webrtc
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