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Unified Diff: webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc

Issue 1307663004: Add a ParseHeader method to RtcpPacket, for parsing common RTCP header. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Merged RtcpPacket header parsing with RTCPUtility Created 5 years, 3 months ago
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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

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