rtcp::Ij renamed to rtcp::ExtendedJitterReport and moved into own file

webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc

Index: webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..deb0dd77ac198a3604862c4872b1e4f4b10e44a5
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2015 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/extended_jitter_report.h"
+
+#include <limits>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+using webrtc::rtcp::RawPacket;
+using webrtc::rtcp::ExtendedJitterReport;
+using webrtc::RTCPUtility::RtcpCommonHeader;
+using webrtc::RTCPUtility::RtcpParseCommonHeader;
+
+namespace webrtc {
+namespace {
+
+class RtcpPacketExtendedJitterReportTest : public ::testing::Test {
+ protected:
+  void BuildPacket() { packet = ij.Build().Pass(); }
+  void ParsePacket() {
+    RtcpCommonHeader header;
+    EXPECT_TRUE(
+        RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header));
+    EXPECT_TRUE(header.packet_type == ExtendedJitterReport::kPacketType);

[åsapersson, 2015/11/12 15:44:31]

Checked within Parse?

[danilchap, 2015/11/13 09:52:48]

Done.

+    EXPECT_EQ(header.BlockSize(), packet->Length());
+    EXPECT_TRUE(parsed_.Parse(
+        header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
+  }
+
+  ExtendedJitterReport ij;
+  rtc::scoped_ptr<RawPacket> packet;
+  const ExtendedJitterReport& parsed() { return parsed_; }
+
+ private:
+  ExtendedJitterReport parsed_;
+};
+
+TEST_F(RtcpPacketExtendedJitterReportTest, NoItem) {
+  // No initialization because packet is empty.
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(0u, parsed().jitters_count());
+}
+
+TEST_F(RtcpPacketExtendedJitterReportTest, OneItem) {
+  EXPECT_TRUE(ij.WithJitter(0x11121314));
+
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(1u, parsed().jitters_count());
+  EXPECT_EQ(0x11121314U, parsed().jitter(0));
+}
+
+TEST_F(RtcpPacketExtendedJitterReportTest, TwoItems) {
+  EXPECT_TRUE(ij.WithJitter(0x11121418));
+  EXPECT_TRUE(ij.WithJitter(0x22242628));
+
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(2u, parsed().jitters_count());
+  EXPECT_EQ(0x11121418U, parsed().jitter(0));
+  EXPECT_EQ(0x22242628U, parsed().jitter(1));
+}
+
+TEST_F(RtcpPacketExtendedJitterReportTest, TooManyItems) {
+  const int kMaxIjItems = (1 << 5) - 1;
+  for (int i = 0; i < kMaxIjItems; ++i) {
+    EXPECT_TRUE(ij.WithJitter(i));
+  }
+  EXPECT_FALSE(ij.WithJitter(kMaxIjItems));
+}
+
+}  // namespace
+}  // namespace webrtc