blob: 58218ddbfd1a1e3b7df960cefb718ada758d30ce [file] [log] [blame]
* 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 "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
using webrtc::rtcp::ReceiverReport;
using webrtc::rtcp::ReportBlock;
namespace webrtc {
const uint32_t kSenderSsrc = 0x12345678;
TEST(RtcpPacketTest, BuildWithTooSmallBuffer) {
ReportBlock rb;
ReceiverReport rr;
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