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Unified Diff: webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request_unittest.cc

Issue 1555543005: [rtp_rtcp] Added Sender Report Request rtcp packet. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 4 years, 11 months ago
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Index: webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..896933d51d78dbe68d92573d18f36bb732fa4dcc
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request_unittest.cc
@@ -0,0 +1,70 @@
+/*
+ * Copyright (c) 2016 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 "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+using testing::ElementsAreArray;
+using testing::make_tuple;
+using webrtc::rtcp::RapidResyncRequest;
+using webrtc::rtcp::RawPacket;
+using webrtc::RTCPUtility::RtcpCommonHeader;
+using webrtc::RTCPUtility::RtcpParseCommonHeader;
+
+namespace webrtc {
+namespace {
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+// Manually created packet matching constants above.
+const uint8_t kPacket[] = {0x85, 205, 0x00, 0x02,
+ 0x12, 0x34, 0x56, 0x78,
+ 0x23, 0x45, 0x67, 0x89};
+const size_t kPacketLength = sizeof(kPacket);
+} // namespace
+
+TEST(RtcpPacketRapidResyncRequestTest, Parse) {
+ RtcpCommonHeader header;
+ ASSERT_TRUE(RtcpParseCommonHeader(kPacket, kPacketLength, &header));
+ RapidResyncRequest mutable_parsed;
+ EXPECT_TRUE(mutable_parsed.Parse(
+ header, kPacket + RtcpCommonHeader::kHeaderSizeBytes));
+ const RapidResyncRequest& parsed = mutable_parsed;
+
+ EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+ EXPECT_EQ(kRemoteSsrc, parsed.media_ssrc());
+}
+
+TEST(RtcpPacketRapidResyncRequestTest, Create) {
+ RapidResyncRequest rrr;
+ rrr.From(kSenderSsrc);
+ rrr.To(kRemoteSsrc);
+
+ rtc::scoped_ptr<RawPacket> packet = rrr.Build();
+
+ EXPECT_THAT(make_tuple(packet->Buffer(), packet->Length()),
+ ElementsAreArray(kPacket));
+}
+
+TEST(RtcpPacketRapidResyncRequestTest, ParseFailsOnWrongSizePacket) {
+ RapidResyncRequest parsed;
+ RtcpCommonHeader header;
+ ASSERT_TRUE(RtcpParseCommonHeader(kPacket, kPacketLength, &header));
+ const size_t kCorrectPayloadSize = header.payload_size_bytes;
+ const uint8_t* payload = kPacket + RtcpCommonHeader::kHeaderSizeBytes;
+
+ header.payload_size_bytes = kCorrectPayloadSize - 1;
+ EXPECT_FALSE(parsed.Parse(header, payload));
+
+ header.payload_size_bytes = kCorrectPayloadSize + 1;
+ EXPECT_FALSE(parsed.Parse(header, payload));
+}
+} // namespace webrtc
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