rtcp::ReportBlock refactored to contain parsing

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

+/*
+ *  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/report_block.h"
+
+#include <limits>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/random.h"
+
+using webrtc::rtcp::ReportBlock;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kRemoteSsrc = 0x23456789;
+const uint8_t kFractionLost = 55;
+// Use values that are streamed differently LE and BE.
+const uint32_t kCumulativeLost = 0x111213;
+const uint32_t kExtHighestSeqNum = 0x22232425;
+const uint32_t kJitter = 0x33343536;
+const uint32_t kLastSr = 0x44454647;
+const uint32_t kDelayLastSr = 0x55565758;
+const size_t kBufferLength = ReportBlock::kLength;
+
+TEST(RtcpPacketReportBlockTest, ParseChecksLength) {
+  uint8_t buffer[kBufferLength];
+  memset(buffer, 0, sizeof(buffer));
+
+  ReportBlock rb;
+  EXPECT_FALSE(rb.Parse(buffer, kBufferLength - 1));
+  EXPECT_TRUE(rb.Parse(buffer, kBufferLength));
+}
+
+TEST(RtcpPacketReportBlockTest, ParseAnyData) {
+  uint8_t buffer[kBufferLength];
+  // Fill buffer with semi-random data.
+  test::Random generator(testing::FLAGS_gtest_random_seed);
+  for (size_t i = 0; i < kBufferLength; ++i)
+    buffer[i] = static_cast<uint8_t>(generator.Rand(0, 0xff));
+
+  ReportBlock rb;
+  EXPECT_TRUE(rb.Parse(buffer, kBufferLength));
+}
+
+TEST(RtcpPacketReportBlockTest, ParseMatchCreate) {
+  ReportBlock rb;
+  rb.To(kRemoteSsrc);
+  rb.WithFractionLost(kFractionLost);
+  rb.WithCumulativeLost(kCumulativeLost);
+  rb.WithExtHighestSeqNum(kExtHighestSeqNum);
+  rb.WithJitter(kJitter);
+  rb.WithLastSr(kLastSr);
+  rb.WithDelayLastSr(kDelayLastSr);
+
+  uint8_t buffer[kBufferLength];
+  rb.Create(buffer);
+
+  ReportBlock parsed;
+  EXPECT_TRUE(parsed.Parse(buffer, kBufferLength));
+
+  EXPECT_EQ(kRemoteSsrc, parsed.source_ssrc());
+  EXPECT_EQ(kFractionLost, parsed.fraction_lost());
+  EXPECT_EQ(kCumulativeLost, parsed.cumulative_lost());
+  EXPECT_EQ(kExtHighestSeqNum, parsed.extended_high_seq_num());
+  EXPECT_EQ(kJitter, parsed.jitter());
+  EXPECT_EQ(kLastSr, parsed.last_sr());
+  EXPECT_EQ(kDelayLastSr, parsed.delay_since_last_sr());
+}
+
+TEST(RtcpPacketReportBlockTest, ValidateCumulativeLost) {
+  const uint32_t kMaxCumulativeLost = 0xffffff;
+  ReportBlock rb;
+  EXPECT_FALSE(rb.WithCumulativeLost(kMaxCumulativeLost + 1));
+  EXPECT_TRUE(rb.WithCumulativeLost(kMaxCumulativeLost));
+}
+
+}  // namespace
+}  // namespace webrtc