webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00

42 lines
1.3 KiB
C++

/*
* Copyright (c) 2014 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 "modules/rtp_rtcp/source/rtcp_packet.h"
#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
#include "test/gmock.h"
#include "test/gtest.h"
using webrtc::rtcp::ReceiverReport;
using webrtc::rtcp::ReportBlock;
namespace webrtc {
const uint32_t kSenderSsrc = 0x12345678;
TEST(RtcpPacketTest, BuildWithTooSmallBuffer) {
ReportBlock rb;
ReceiverReport rr;
rr.SetSenderSsrc(kSenderSsrc);
EXPECT_TRUE(rr.AddReportBlock(rb));
const size_t kRrLength = 8;
const size_t kReportBlockLength = 24;
// No packet.
class Verifier : public rtcp::RtcpPacket::PacketReadyCallback {
void OnPacketReady(uint8_t* data, size_t length) override {
ADD_FAILURE() << "Packet should not fit within max size.";
}
} verifier;
const size_t kBufferSize = kRrLength + kReportBlockLength - 1;
uint8_t buffer[kBufferSize];
EXPECT_FALSE(rr.BuildExternalBuffer(buffer, kBufferSize, &verifier));
}
} // namespace webrtc