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..3c3adb510dbaee8f322949357c24d95ab5a786c2 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 +45,87 @@ 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; |
åsapersson
2015/09/08 10:20:41
kHeaderSize -> kHeaderSizeInBytes
kBufferCapacity
sprang_webrtc
2015/09/09 09:18:53
Done.
|
+ uint8_t buffer[kBufferCapacity]; |
+ RTCPUtility::RtcpCommonHeader header; |
+}; |
+ |
åsapersson
2015/09/08 10:20:41
I think these parsing tests should be where the pa
sprang_webrtc
2015/09/09 09:18:53
Fair enough. My long (long) term hope is to rewrit
|
+TEST_F(RtcpPacketHeaderTest, HeaderSize) { |
åsapersson
2015/09/08 10:20:41
Maybe rename to something like TooShortHeaderSize.
sprang_webrtc
2015/09/09 09:18:54
Renamed to TooSmallBuffer.
|
+ // Buffer needs to be able to hold the header. |
+ for (size_t i = 0; i < kHeaderSize; ++i) |
+ EXPECT_FALSE(RtcpParseCommonHeader(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, RtcpParseCommonHeader(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(RtcpParseCommonHeader(buffer, 0, kSizeInBytes - 1, &header)); |
+ EXPECT_TRUE(RtcpParseCommonHeader(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; |
åsapersson
2015/09/08 10:20:41
kPayloadSize -> kPayloadSizeInBytes
sprang_webrtc
2015/09/09 09:18:53
Done.
|
+ |
+ // Padding one byte larger than possible. |
+ buffer[kSizeInBytes - 1] = kPayloadSize + 1; |
+ EXPECT_FALSE(RtcpParseCommonHeader(buffer, 0, kSizeInBytes, &header)); |
+ |
+ // Pure padding packet? |
+ buffer[kSizeInBytes - 1] = kPayloadSize; |
+ EXPECT_TRUE(RtcpParseCommonHeader(buffer, 0, kSizeInBytes, &header)); |
+ EXPECT_EQ(0u, header.payload_size_bytes); |
+ |
+ // Single byte of actual data. |
+ buffer[kSizeInBytes - 1] = kPayloadSize - 1; |
+ EXPECT_TRUE(RtcpParseCommonHeader(buffer, 0, kSizeInBytes, &header)); |
+ EXPECT_EQ(1u, header.payload_size_bytes); |
+} |
+ |
+TEST_F(RtcpPacketHeaderTest, FormatAndPayloadType) {} |
åsapersson
2015/09/08 10:20:41
remove
sprang_webrtc
2015/09/09 09:18:53
Or implement :)
|
+ |
TEST(RtcpPacketTest, Rr) { |
ReceiverReport rr; |
rr.From(kSenderSsrc); |
@@ -1085,4 +1164,5 @@ TEST(RtcpPacketTest, XrWithTooManyBlocks) { |
EXPECT_FALSE(xr.WithVoipMetric(&voip_metric)); |
} |
+} // namespace rtcp |
} // namespace webrtc |