| Index: webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
|
| index 77520b633bc9de3eced7e4824137f3b320c03844..7ad99780177c76be1461c44c80db0faf414b48cc 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
|
| @@ -13,6 +13,8 @@
|
| #include "testing/gmock/include/gmock/gmock.h"
|
| #include "testing/gtest/include/gtest/gtest.h"
|
|
|
| +#include "webrtc/base/checks.h"
|
| +#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
|
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
|
| #include "webrtc/test/rtcp_packet_parser.h"
|
|
|
| @@ -43,10 +45,93 @@ using webrtc::rtcp::Xr;
|
| using webrtc::test::RtcpPacketParser;
|
|
|
| namespace webrtc {
|
| +namespace rtcp {
|
|
|
| const uint32_t kSenderSsrc = 0x12345678;
|
| const uint32_t kRemoteSsrc = 0x23456789;
|
|
|
| +// Override RtcpPacket so we can test protected methods.
|
| +class RtcpPacketHeaderTest : public ::testing::Test,
|
| + protected RtcpPacket {
|
| + public:
|
| + RtcpPacketHeaderTest() {
|
| + memset(buffer, 0, kBufferCapacity);
|
| + }
|
| +
|
| + virtual ~RtcpPacketHeaderTest() {}
|
| +
|
| + bool Create(uint8_t* packet,
|
| + size_t* index,
|
| + size_t max_length,
|
| + PacketReadyCallback* callback) const override {
|
| + RTC_NOTREACHED();
|
| + return false;
|
| + }
|
| +
|
| + size_t BlockLength() const { return 0; }
|
| +
|
| + protected:
|
| + static const size_t kHeaderSize = 4;
|
| + static const size_t kBufferCapacity = 40;
|
| + uint8_t buffer[kBufferCapacity];
|
| + RtcpPacket::CommonHeader header;
|
| +};
|
| +
|
| +TEST_F(RtcpPacketHeaderTest, HeaderSize) {
|
| + // Buffer needs to be able to hold the header.
|
| + for (size_t i = 0; i < kHeaderSize; ++i)
|
| + EXPECT_FALSE(ParseHeader(buffer, 0, i, &header));
|
| +}
|
| +
|
| +TEST_F(RtcpPacketHeaderTest, Version) {
|
| + // Version 2 is the only allowed for now.
|
| + for (int v = 0; v < 4; ++v) {
|
| + buffer[0] = v << 6;
|
| + EXPECT_EQ(v == 2, ParseHeader(buffer, 0, kHeaderSize, &header));
|
| + }
|
| +}
|
| +
|
| +TEST_F(RtcpPacketHeaderTest, PacketSize) {
|
| + // Set v = 2, leave p, fmt, pt as 0.
|
| + buffer[0] = 2 << 6;
|
| +
|
| + const size_t kBlockSize = 3;
|
| + ByteWriter<uint16_t>::WriteBigEndian(&buffer[2], kBlockSize);
|
| + const size_t kSizeInBytes = (kBlockSize + 1) * 4;
|
| +
|
| + EXPECT_FALSE(ParseHeader(buffer, 0, kSizeInBytes - 1, &header));
|
| + EXPECT_TRUE(ParseHeader(buffer, 0, kSizeInBytes, &header));
|
| +}
|
| +
|
| +TEST_F(RtcpPacketHeaderTest, PayloadSize) {
|
| + // Set v = 2, p = 1, but leave fmt, pt as 0.
|
| + buffer[0] = (2 << 6) | (1 << 5);
|
| +
|
| + const size_t kBlockSize = 3;
|
| + ByteWriter<uint16_t>::WriteBigEndian(&buffer[2], kBlockSize);
|
| + const size_t kSizeInBytes = (kBlockSize + 1) * 4;
|
| + const size_t kPayloadSize = kSizeInBytes - kHeaderSize;
|
| +
|
| + // Padding one byte larger than possible.
|
| + buffer[kSizeInBytes - 1] = kPayloadSize + 1;
|
| + EXPECT_FALSE(ParseHeader(buffer, 0, kSizeInBytes, &header));
|
| +
|
| + // Pure padding packet?
|
| + buffer[kSizeInBytes - 1] = kPayloadSize;
|
| + EXPECT_TRUE(ParseHeader(buffer, 0, kSizeInBytes, &header));
|
| + EXPECT_EQ(0u, header.payload_size_bytes);
|
| +
|
| + // Single byte of actual data.
|
| + buffer[kSizeInBytes - 1] = kPayloadSize - 1;
|
| + EXPECT_TRUE(ParseHeader(buffer, 0, kSizeInBytes, &header));
|
| + EXPECT_EQ(1u, header.payload_size_bytes);
|
| +
|
| +}
|
| +
|
| +TEST_F(RtcpPacketHeaderTest, FormatAndPayloadType) {
|
| +
|
| +}
|
| +
|
| TEST(RtcpPacketTest, Rr) {
|
| ReceiverReport rr;
|
| rr.From(kSenderSsrc);
|
| @@ -1085,4 +1170,5 @@ TEST(RtcpPacketTest, XrWithTooManyBlocks) {
|
| EXPECT_FALSE(xr.WithVoipMetric(&voip_metric));
|
| }
|
|
|
| +} // namespace rtcp
|
| } // namespace webrtc
|
|
|