rtcp::ReceiverReport moved into own file and got Parse function
BUG=webrtc:5260
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1453083002 .
Cr-Commit-Position: refs/heads/master@{#10897}
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index 3d99253..5d7899c 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -308,6 +308,7 @@
'rtp_rtcp/source/rtcp_packet/bye_unittest.cc',
'rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc',
'rtp_rtcp/source/rtcp_packet/pli_unittest.cc',
+ 'rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc',
'rtp_rtcp/source/rtcp_packet/report_block_unittest.cc',
'rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc',
'rtp_rtcp/source/rtcp_receiver_unittest.cc',
diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn
index 9624658..21869b4 100644
--- a/webrtc/modules/rtp_rtcp/BUILD.gn
+++ b/webrtc/modules/rtp_rtcp/BUILD.gn
@@ -56,6 +56,8 @@
"source/rtcp_packet/pli.h",
"source/rtcp_packet/psfb.cc",
"source/rtcp_packet/psfb.h",
+ "source/rtcp_packet/receiver_report.cc",
+ "source/rtcp_packet/receiver_report.h",
"source/rtcp_packet/report_block.cc",
"source/rtcp_packet/report_block.h",
"source/rtcp_packet/transport_feedback.cc",
diff --git a/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi b/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
index 1e69c12..a315b02 100644
--- a/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
+++ b/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
@@ -51,6 +51,8 @@
'source/rtcp_packet/pli.h',
'source/rtcp_packet/psfb.cc',
'source/rtcp_packet/psfb.h',
+ 'source/rtcp_packet/receiver_report.cc',
+ 'source/rtcp_packet/receiver_report.h',
'source/rtcp_packet/report_block.cc',
'source/rtcp_packet/report_block.h',
'source/rtcp_packet/transport_feedback.cc',
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
index d23abdc..41ebcad 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
@@ -21,7 +21,6 @@
using webrtc::RTCPUtility::PT_APP;
using webrtc::RTCPUtility::PT_IJ;
using webrtc::RTCPUtility::PT_PSFB;
-using webrtc::RTCPUtility::PT_RR;
using webrtc::RTCPUtility::PT_RTPFB;
using webrtc::RTCPUtility::PT_SDES;
using webrtc::RTCPUtility::PT_SR;
@@ -36,7 +35,6 @@
using webrtc::RTCPUtility::RTCPPacketPSFBSLI;
using webrtc::RTCPUtility::RTCPPacketPSFBSLIItem;
using webrtc::RTCPUtility::RTCPPacketReportBlockItem;
-using webrtc::RTCPUtility::RTCPPacketRR;
using webrtc::RTCPUtility::RTCPPacketRTPFBNACK;
using webrtc::RTCPUtility::RTCPPacketRTPFBNACKItem;
using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBN;
@@ -119,21 +117,6 @@
AssignUWord32(buffer, pos, sr.SenderOctetCount);
}
-// Receiver report (RR), header (RFC 3550).
-//
-// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// |V=2|P| RC | PT=RR=201 | length |
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// | SSRC of packet sender |
-// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-
-void CreateReceiverReport(const RTCPPacketRR& rr,
- uint8_t* buffer,
- size_t* pos) {
- AssignUWord32(buffer, pos, rr.SenderSSRC);
-}
-
// Report block (RFC 3550).
//
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
@@ -681,30 +664,6 @@
return true;
}
-bool ReceiverReport::Create(uint8_t* packet,
- size_t* index,
- size_t max_length,
- RtcpPacket::PacketReadyCallback* callback) const {
- while (*index + BlockLength() > max_length) {
- if (!OnBufferFull(packet, index, callback))
- return false;
- }
- CreateHeader(rr_.NumberOfReportBlocks, PT_RR, HeaderLength(), packet, index);
- CreateReceiverReport(rr_, packet, index);
- CreateReportBlocks(report_blocks_, packet, index);
- return true;
-}
-
-bool ReceiverReport::WithReportBlock(const ReportBlock& block) {
- if (report_blocks_.size() >= kMaxNumberOfReportBlocks) {
- LOG(LS_WARNING) << "Max report blocks reached.";
- return false;
- }
- report_blocks_.push_back(block);
- rr_.NumberOfReportBlocks = report_blocks_.size();
- return true;
-}
-
bool Sdes::Create(uint8_t* packet,
size_t* index,
size_t max_length,
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
index 39c2728..1f5e223 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
@@ -216,52 +216,6 @@
RTC_DISALLOW_COPY_AND_ASSIGN(SenderReport);
};
-//
-// RTCP receiver report (RFC 3550).
-//
-// 0 1 2 3
-// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// |V=2|P| RC | PT=RR=201 | length |
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// | SSRC of packet sender |
-// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-// | report block(s) |
-// | .... |
-
-class ReceiverReport : public RtcpPacket {
- public:
- ReceiverReport() : RtcpPacket() {
- memset(&rr_, 0, sizeof(rr_));
- }
-
- virtual ~ReceiverReport() {}
-
- void From(uint32_t ssrc) {
- rr_.SenderSSRC = ssrc;
- }
- bool WithReportBlock(const ReportBlock& block);
-
- protected:
- bool Create(uint8_t* packet,
- size_t* index,
- size_t max_length,
- RtcpPacket::PacketReadyCallback* callback) const override;
-
- private:
- static const int kMaxNumberOfReportBlocks = 0x1F;
-
- size_t BlockLength() const {
- const size_t kRrHeaderLength = 8;
- return kRrHeaderLength + report_blocks_.size() * kReportBlockLength;
- }
-
- RTCPUtility::RTCPPacketRR rr_;
- std::vector<ReportBlock> report_blocks_;
-
- RTC_DISALLOW_COPY_AND_ASSIGN(ReceiverReport);
-};
-
// Source Description (SDES) (RFC 3550).
//
// 0 1 2 3
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.cc
new file mode 100644
index 0000000..ef64b4f
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.cc
@@ -0,0 +1,89 @@
+/*
+ * 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/receiver_report.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::RTCPUtility::RtcpCommonHeader;
+
+namespace webrtc {
+namespace rtcp {
+
+//
+// RTCP receiver report (RFC 3550).
+//
+// 0 1 2 3
+// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |V=2|P| RC | PT=RR=201 | length |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// | SSRC of packet sender |
+// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+// | report block(s) |
+// | .... |
+bool ReceiverReport::Parse(const RTCPUtility::RtcpCommonHeader& header,
+ const uint8_t* payload) {
+ RTC_DCHECK(header.packet_type == kPacketType);
+
+ const uint8_t report_blocks_count = header.count_or_format;
+
+ if (header.payload_size_bytes <
+ kRrBaseLength + report_blocks_count * ReportBlock::kLength) {
+ LOG(LS_WARNING) << "Packet is too small to contain all the data.";
+ return false;
+ }
+
+ sender_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(payload);
+
+ const uint8_t* next_report_block = payload + kRrBaseLength;
+
+ report_blocks_.resize(report_blocks_count);
+ for (ReportBlock& block : report_blocks_) {
+ block.Parse(next_report_block, ReportBlock::kLength);
+ next_report_block += ReportBlock::kLength;
+ }
+
+ RTC_DCHECK_LE(next_report_block, payload + header.payload_size_bytes);
+ return true;
+}
+
+bool ReceiverReport::Create(uint8_t* packet,
+ size_t* index,
+ size_t max_length,
+ RtcpPacket::PacketReadyCallback* callback) const {
+ while (*index + BlockLength() > max_length) {
+ if (!OnBufferFull(packet, index, callback))
+ return false;
+ }
+ CreateHeader(report_blocks_.size(), kPacketType, HeaderLength(), packet,
+ index);
+ ByteWriter<uint32_t>::WriteBigEndian(packet + *index, sender_ssrc_);
+ *index += kRrBaseLength;
+ for (const ReportBlock& block : report_blocks_) {
+ block.Create(packet + *index);
+ *index += ReportBlock::kLength;
+ }
+ return true;
+}
+
+bool ReceiverReport::WithReportBlock(const ReportBlock& block) {
+ if (report_blocks_.size() >= kMaxNumberOfReportBlocks) {
+ LOG(LS_WARNING) << "Max report blocks reached.";
+ return false;
+ }
+ report_blocks_.push_back(block);
+ return true;
+}
+
+} // namespace rtcp
+} // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h
new file mode 100644
index 0000000..172a84e
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h
@@ -0,0 +1,66 @@
+/*
+ * 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.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RECEIVER_REPORT_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RECEIVER_REPORT_H_
+
+#include <vector>
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class ReceiverReport : public RtcpPacket {
+ public:
+ static const uint8_t kPacketType = 201;
+ ReceiverReport() : sender_ssrc_(0) {}
+
+ virtual ~ReceiverReport() {}
+
+ // Parse assumes header is already parsed and validated.
+ bool Parse(const RTCPUtility::RtcpCommonHeader& header,
+ const uint8_t* payload); // Size of the payload is in the header.
+
+ void From(uint32_t ssrc) { sender_ssrc_ = ssrc; }
+ bool WithReportBlock(const ReportBlock& block);
+
+ uint32_t sender_ssrc() const { return sender_ssrc_; }
+ const std::vector<ReportBlock>& report_blocks() const {
+ return report_blocks_;
+ }
+
+ protected:
+ bool Create(uint8_t* packet,
+ size_t* index,
+ size_t max_length,
+ RtcpPacket::PacketReadyCallback* callback) const override;
+
+ private:
+ static const size_t kRrBaseLength = 4;
+ static const size_t kMaxNumberOfReportBlocks = 0x1F;
+
+ size_t BlockLength() const {
+ return kHeaderLength + kRrBaseLength +
+ report_blocks_.size() * ReportBlock::kLength;
+ }
+
+ uint32_t sender_ssrc_;
+ std::vector<ReportBlock> report_blocks_;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(ReceiverReport);
+};
+
+} // namespace rtcp
+} // namespace webrtc
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RECEIVER_REPORT_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc
new file mode 100644
index 0000000..4fd329f
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc
@@ -0,0 +1,145 @@
+/*
+ * 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/receiver_report.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+using webrtc::rtcp::RawPacket;
+using webrtc::rtcp::ReceiverReport;
+using webrtc::rtcp::ReportBlock;
+using webrtc::RTCPUtility::RtcpCommonHeader;
+using webrtc::RTCPUtility::RtcpParseCommonHeader;
+
+namespace webrtc {
+namespace {
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+const uint8_t kFractionLost = 55;
+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;
+// Manually created ReceiverReport with one ReportBlock matching constants
+// above.
+// Having this block allows to test Create and Parse separately.
+const uint8_t kPacket[] = {0x81, 201, 0x00, 0x07, 0x12, 0x34, 0x56, 0x78,
+ 0x23, 0x45, 0x67, 0x89, 55, 0x11, 0x12, 0x13,
+ 0x22, 0x23, 0x24, 0x25, 0x33, 0x34, 0x35, 0x36,
+ 0x44, 0x45, 0x46, 0x47, 0x55, 0x56, 0x57, 0x58};
+const size_t kPacketLength = sizeof(kPacket);
+
+class RtcpPacketReceiverReportTest : public ::testing::Test {
+ protected:
+ void BuildPacket() { packet = rr.Build().Pass(); }
+ void ParsePacket() {
+ RtcpCommonHeader header;
+ EXPECT_TRUE(
+ RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header));
+ EXPECT_EQ(header.BlockSize(), packet->Length());
+ EXPECT_TRUE(parsed_.Parse(
+ header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
+ }
+
+ ReceiverReport rr;
+ rtc::scoped_ptr<RawPacket> packet;
+ const ReceiverReport& parsed() { return parsed_; }
+
+ private:
+ ReceiverReport parsed_;
+};
+
+TEST_F(RtcpPacketReceiverReportTest, Parse) {
+ RtcpCommonHeader header;
+ RtcpParseCommonHeader(kPacket, kPacketLength, &header);
+ EXPECT_TRUE(rr.Parse(header, kPacket + RtcpCommonHeader::kHeaderSizeBytes));
+ const ReceiverReport& parsed = rr;
+
+ EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+ EXPECT_EQ(1u, parsed.report_blocks().size());
+ const ReportBlock& rb = parsed.report_blocks().front();
+ EXPECT_EQ(kRemoteSsrc, rb.source_ssrc());
+ EXPECT_EQ(kFractionLost, rb.fraction_lost());
+ EXPECT_EQ(kCumulativeLost, rb.cumulative_lost());
+ EXPECT_EQ(kExtHighestSeqNum, rb.extended_high_seq_num());
+ EXPECT_EQ(kJitter, rb.jitter());
+ EXPECT_EQ(kLastSr, rb.last_sr());
+ EXPECT_EQ(kDelayLastSr, rb.delay_since_last_sr());
+}
+
+TEST_F(RtcpPacketReceiverReportTest, ParseFailsOnIncorrectSize) {
+ RtcpCommonHeader header;
+ RtcpParseCommonHeader(kPacket, kPacketLength, &header);
+ header.count_or_format++; // Damage the packet.
+ EXPECT_FALSE(rr.Parse(header, kPacket + RtcpCommonHeader::kHeaderSizeBytes));
+}
+
+TEST_F(RtcpPacketReceiverReportTest, Create) {
+ rr.From(kSenderSsrc);
+ ReportBlock rb;
+ rb.To(kRemoteSsrc);
+ rb.WithFractionLost(kFractionLost);
+ rb.WithCumulativeLost(kCumulativeLost);
+ rb.WithExtHighestSeqNum(kExtHighestSeqNum);
+ rb.WithJitter(kJitter);
+ rb.WithLastSr(kLastSr);
+ rb.WithDelayLastSr(kDelayLastSr);
+ rr.WithReportBlock(rb);
+
+ BuildPacket();
+
+ ASSERT_EQ(kPacketLength, packet->Length());
+ EXPECT_EQ(0, memcmp(kPacket, packet->Buffer(), kPacketLength));
+}
+
+TEST_F(RtcpPacketReceiverReportTest, WithoutReportBlocks) {
+ rr.From(kSenderSsrc);
+
+ BuildPacket();
+ ParsePacket();
+
+ EXPECT_EQ(kSenderSsrc, parsed().sender_ssrc());
+ EXPECT_EQ(0u, parsed().report_blocks().size());
+}
+
+TEST_F(RtcpPacketReceiverReportTest, WithTwoReportBlocks) {
+ ReportBlock rb1;
+ rb1.To(kRemoteSsrc);
+ ReportBlock rb2;
+ rb2.To(kRemoteSsrc + 1);
+
+ rr.From(kSenderSsrc);
+ EXPECT_TRUE(rr.WithReportBlock(rb1));
+ EXPECT_TRUE(rr.WithReportBlock(rb2));
+
+ BuildPacket();
+ ParsePacket();
+
+ EXPECT_EQ(kSenderSsrc, parsed().sender_ssrc());
+ EXPECT_EQ(2u, parsed().report_blocks().size());
+ EXPECT_EQ(kRemoteSsrc, parsed().report_blocks()[0].source_ssrc());
+ EXPECT_EQ(kRemoteSsrc + 1, parsed().report_blocks()[1].source_ssrc());
+}
+
+TEST_F(RtcpPacketReceiverReportTest, WithTooManyReportBlocks) {
+ rr.From(kSenderSsrc);
+ const size_t kMaxReportBlocks = (1 << 5) - 1;
+ ReportBlock rb;
+ for (size_t i = 0; i < kMaxReportBlocks; ++i) {
+ rb.To(kRemoteSsrc + i);
+ EXPECT_TRUE(rr.WithReportBlock(rb));
+ }
+ rb.To(kRemoteSsrc + kMaxReportBlocks);
+ EXPECT_FALSE(rr.WithReportBlock(rb));
+}
+
+} // namespace
+} // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
index 1743d99..dc2b113 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
@@ -16,6 +16,7 @@
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
#include "webrtc/test/rtcp_packet_parser.h"
using ::testing::ElementsAre;
@@ -26,16 +27,15 @@
using webrtc::rtcp::Empty;
using webrtc::rtcp::Fir;
using webrtc::rtcp::Nack;
-using webrtc::rtcp::Sdes;
-using webrtc::rtcp::SenderReport;
-using webrtc::rtcp::Sli;
using webrtc::rtcp::RawPacket;
using webrtc::rtcp::ReceiverReport;
using webrtc::rtcp::Remb;
using webrtc::rtcp::ReportBlock;
using webrtc::rtcp::Rpsi;
using webrtc::rtcp::Rrtr;
+using webrtc::rtcp::Sdes;
using webrtc::rtcp::SenderReport;
+using webrtc::rtcp::Sli;
using webrtc::rtcp::Tmmbn;
using webrtc::rtcp::Tmmbr;
using webrtc::rtcp::VoipMetric;
@@ -47,81 +47,6 @@
const uint32_t kSenderSsrc = 0x12345678;
const uint32_t kRemoteSsrc = 0x23456789;
-TEST(RtcpPacketTest, Rr) {
- ReceiverReport rr;
- rr.From(kSenderSsrc);
-
- rtc::scoped_ptr<RawPacket> packet(rr.Build());
- RtcpPacketParser parser;
- parser.Parse(packet->Buffer(), packet->Length());
- EXPECT_EQ(1, parser.receiver_report()->num_packets());
- EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc());
- EXPECT_EQ(0, parser.report_block()->num_packets());
-}
-
-TEST(RtcpPacketTest, RrWithOneReportBlock) {
- ReportBlock rb;
- rb.To(kRemoteSsrc);
- rb.WithFractionLost(55);
- rb.WithCumulativeLost(0x111111);
- rb.WithExtHighestSeqNum(0x22222222);
- rb.WithJitter(0x33333333);
- rb.WithLastSr(0x44444444);
- rb.WithDelayLastSr(0x55555555);
-
- ReceiverReport rr;
- rr.From(kSenderSsrc);
- EXPECT_TRUE(rr.WithReportBlock(rb));
-
- rtc::scoped_ptr<RawPacket> packet(rr.Build());
- RtcpPacketParser parser;
- parser.Parse(packet->Buffer(), packet->Length());
- EXPECT_EQ(1, parser.receiver_report()->num_packets());
- EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc());
- EXPECT_EQ(1, parser.report_block()->num_packets());
- EXPECT_EQ(kRemoteSsrc, parser.report_block()->Ssrc());
- EXPECT_EQ(55U, parser.report_block()->FractionLost());
- EXPECT_EQ(0x111111U, parser.report_block()->CumPacketLost());
- EXPECT_EQ(0x22222222U, parser.report_block()->ExtHighestSeqNum());
- EXPECT_EQ(0x33333333U, parser.report_block()->Jitter());
- EXPECT_EQ(0x44444444U, parser.report_block()->LastSr());
- EXPECT_EQ(0x55555555U, parser.report_block()->DelayLastSr());
-}
-
-TEST(RtcpPacketTest, RrWithTwoReportBlocks) {
- ReportBlock rb1;
- rb1.To(kRemoteSsrc);
- ReportBlock rb2;
- rb2.To(kRemoteSsrc + 1);
-
- ReceiverReport rr;
- rr.From(kSenderSsrc);
- EXPECT_TRUE(rr.WithReportBlock(rb1));
- EXPECT_TRUE(rr.WithReportBlock(rb2));
-
- rtc::scoped_ptr<RawPacket> packet(rr.Build());
- RtcpPacketParser parser;
- parser.Parse(packet->Buffer(), packet->Length());
- EXPECT_EQ(1, parser.receiver_report()->num_packets());
- EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc());
- EXPECT_EQ(2, parser.report_block()->num_packets());
- EXPECT_EQ(1, parser.report_blocks_per_ssrc(kRemoteSsrc));
- EXPECT_EQ(1, parser.report_blocks_per_ssrc(kRemoteSsrc + 1));
-}
-
-TEST(RtcpPacketTest, RrWithTooManyReportBlocks) {
- ReceiverReport rr;
- rr.From(kSenderSsrc);
- const int kMaxReportBlocks = (1 << 5) - 1;
- ReportBlock rb;
- for (int i = 0; i < kMaxReportBlocks; ++i) {
- rb.To(kRemoteSsrc + i);
- EXPECT_TRUE(rr.WithReportBlock(rb));
- }
- rb.To(kRemoteSsrc + kMaxReportBlocks);
- EXPECT_FALSE(rr.WithReportBlock(rb));
-}
-
TEST(RtcpPacketTest, Sr) {
SenderReport sr;
sr.From(kSenderSsrc);
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
index ac0389a..b75fb73 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
@@ -29,6 +29,7 @@
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index 5add7cf..ac3f337 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -25,6 +25,7 @@
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
