diff options
Diffstat (limited to 'webrtc/modules')
| -rw-r--r-- | webrtc/modules/modules.gyp | 1 | ||||
| -rw-r--r-- | webrtc/modules/rtp_rtcp/BUILD.gn | 2 | ||||
| -rw-r--r-- | webrtc/modules/rtp_rtcp/rtp_rtcp.gypi | 2 | ||||
| -rw-r--r-- | webrtc/modules/rtp_rtcp/source/rtcp_packet.cc | 47 | ||||
| -rw-r--r-- | webrtc/modules/rtp_rtcp/source/rtcp_packet.h | 50 | ||||
| -rw-r--r-- | webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc | 133 | ||||
| -rw-r--r-- | webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h | 63 | ||||
| -rw-r--r-- | webrtc/modules/rtp_rtcp/source/rtcp_packet/bye_unittest.cc | 173 | ||||
| -rw-r--r-- | webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc | 38 | ||||
| -rw-r--r-- | webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc | 1 | ||||
| -rw-r--r-- | webrtc/modules/rtp_rtcp/source/rtcp_sender.cc | 1 |
11 files changed, 377 insertions, 134 deletions
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp index 44cf73e05c..0c981e691f 100644 --- a/webrtc/modules/modules.gyp +++ b/webrtc/modules/modules.gyp @@ -305,6 +305,7 @@ 'rtp_rtcp/source/rtcp_format_remb_unittest.cc', 'rtp_rtcp/source/rtcp_packet_unittest.cc', 'rtp_rtcp/source/rtcp_packet/app_unittest.cc', + 'rtp_rtcp/source/rtcp_packet/bye_unittest.cc', 'rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc', 'rtp_rtcp/source/rtcp_packet/report_block_unittest.cc', 'rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc', diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index f7d887d14c..03b72adf6f 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -48,6 +48,8 @@ source_set("rtp_rtcp") { "source/rtcp_packet.h", "source/rtcp_packet/app.cc", "source/rtcp_packet/app.h", + "source/rtcp_packet/bye.cc", + "source/rtcp_packet/bye.h", "source/rtcp_packet/extended_jitter_report.cc", "source/rtcp_packet/extended_jitter_report.h", "source/rtcp_packet/report_block.cc", diff --git a/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi b/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi index 71e2ce7d69..eaa5b157fb 100644 --- a/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi +++ b/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi @@ -43,6 +43,8 @@ 'source/rtcp_packet.h', 'source/rtcp_packet/app.cc', 'source/rtcp_packet/app.h', + 'source/rtcp_packet/bye.cc', + 'source/rtcp_packet/bye.h', 'source/rtcp_packet/extended_jitter_report.cc', 'source/rtcp_packet/extended_jitter_report.h', 'source/rtcp_packet/report_block.cc', diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc index be03d9b41a..e3305186a0 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc +++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc @@ -19,7 +19,6 @@ using webrtc::RTCPUtility::kBtReceiverReferenceTime; using webrtc::RTCPUtility::kBtVoipMetric; using webrtc::RTCPUtility::PT_APP; -using webrtc::RTCPUtility::PT_BYE; using webrtc::RTCPUtility::PT_IJ; using webrtc::RTCPUtility::PT_PSFB; using webrtc::RTCPUtility::PT_RR; @@ -29,7 +28,6 @@ using webrtc::RTCPUtility::PT_SR; using webrtc::RTCPUtility::PT_XR; using webrtc::RTCPUtility::RTCPPacketAPP; -using webrtc::RTCPUtility::RTCPPacketBYE; using webrtc::RTCPUtility::RTCPPacketPSFBAPP; using webrtc::RTCPUtility::RTCPPacketPSFBFIR; using webrtc::RTCPUtility::RTCPPacketPSFBFIRItem; @@ -205,28 +203,6 @@ void CreateSdes(const std::vector<Sdes::Chunk>& chunks, } } -// Bye packet (BYE) (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| SC | PT=BYE=203 | length | -// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -// | SSRC/CSRC | -// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -// : ... : -// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ -// (opt) | length | reason for leaving ... -// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - -void CreateBye(const RTCPPacketBYE& bye, - const std::vector<uint32_t>& csrcs, - uint8_t* buffer, - size_t* pos) { - AssignUWord32(buffer, pos, bye.SenderSSRC); - for (uint32_t csrc : csrcs) - AssignUWord32(buffer, pos, csrc); -} - // RFC 4585: Feedback format. // // Common packet format: @@ -801,29 +777,6 @@ size_t Sdes::BlockLength() const { return length; } -bool Bye::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; - } - size_t length = HeaderLength(); - CreateHeader(length, PT_BYE, length, packet, index); - CreateBye(bye_, csrcs_, packet, index); - return true; -} - -bool Bye::WithCsrc(uint32_t csrc) { - if (csrcs_.size() >= kMaxNumberOfCsrcs) { - LOG(LS_WARNING) << "Max CSRC size reached."; - return false; - } - csrcs_.push_back(csrc); - return true; -} - bool Pli::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 bea83070cb..3ac66184bd 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_packet.h +++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet.h @@ -319,56 +319,6 @@ class Sdes : public RtcpPacket { RTC_DISALLOW_COPY_AND_ASSIGN(Sdes); }; -// -// Bye packet (BYE) (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| SC | PT=BYE=203 | length | -// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -// | SSRC/CSRC | -// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ -// : ... : -// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ -// (opt) | length | reason for leaving ... -// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - -class Bye : public RtcpPacket { - public: - Bye() : RtcpPacket() { - memset(&bye_, 0, sizeof(bye_)); - } - - virtual ~Bye() {} - - void From(uint32_t ssrc) { - bye_.SenderSSRC = ssrc; - } - - bool WithCsrc(uint32_t csrc); - - // TODO(sprang): Add support for reason field? - - protected: - bool Create(uint8_t* packet, - size_t* index, - size_t max_length, - RtcpPacket::PacketReadyCallback* callback) const override; - - private: - static const int kMaxNumberOfCsrcs = 0x1f - 1; // First item is sender SSRC. - - size_t BlockLength() const { - size_t source_count = 1 + csrcs_.size(); - return kHeaderLength + 4 * source_count; - } - - RTCPUtility::RTCPPacketBYE bye_; - std::vector<uint32_t> csrcs_; - - RTC_DISALLOW_COPY_AND_ASSIGN(Bye); -}; - // RFC 4585: Feedback format. // // Common packet format: diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc new file mode 100644 index 0000000000..4cfc921ce5 --- /dev/null +++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc @@ -0,0 +1,133 @@ +/* + * 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/bye.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 { + +// Bye packet (BYE) (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| SC | PT=BYE=203 | length | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// | SSRC/CSRC | +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +// : ... : +// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ +// (opt) | length | reason for leaving ... +// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +Bye::Bye() : sender_ssrc_(0) {} + +bool Bye::Parse(const RtcpCommonHeader& header, const uint8_t* payload) { + RTC_DCHECK(header.packet_type == kPacketType); + + const uint8_t src_count = header.count_or_format; + // Validate packet. + if (header.payload_size_bytes < 4u * src_count) { + LOG(LS_WARNING) + << "Packet is too small to contain CSRCs it promise to have."; + return false; + } + bool has_reason = (header.payload_size_bytes > 4u * src_count); + uint8_t reason_length = 0; + if (has_reason) { + reason_length = payload[4u * src_count]; + if (header.payload_size_bytes - 4u * src_count < 1u + reason_length) { + LOG(LS_WARNING) << "Invalid reason length: " << reason_length; + return false; + } + } + // Once sure packet is valid, copy values. + if (src_count == 0) { // A count value of zero is valid, but useless. + sender_ssrc_ = 0; + csrcs_.clear(); + } else { + sender_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(payload); + csrcs_.resize(src_count - 1); + for (size_t i = 1; i < src_count; ++i) + csrcs_[i - 1] = ByteReader<uint32_t>::ReadBigEndian(&payload[4 * i]); + } + + if (has_reason) { + reason_.assign(reinterpret_cast<const char*>(&payload[4u * src_count + 1]), + reason_length); + } else { + reason_.clear(); + } + + return true; +} + +bool Bye::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; + } + const size_t index_end = *index + BlockLength(); + + CreateHeader(1 + csrcs_.size(), kPacketType, HeaderLength(), packet, index); + // Store srcs of the leaving clients. + ByteWriter<uint32_t>::WriteBigEndian(&packet[*index], sender_ssrc_); + *index += sizeof(uint32_t); + for (uint32_t csrc : csrcs_) { + ByteWriter<uint32_t>::WriteBigEndian(&packet[*index], csrc); + *index += sizeof(uint32_t); + } + // Store the reason to leave. + if (!reason_.empty()) { + uint8_t reason_length = reason_.size(); + packet[(*index)++] = reason_length; + memcpy(&packet[*index], reason_.data(), reason_length); + *index += reason_length; + // Add padding bytes if needed. + size_t bytes_to_pad = index_end - *index; + RTC_DCHECK_LE(bytes_to_pad, 3u); + if (bytes_to_pad > 0) { + memset(&packet[*index], 0, bytes_to_pad); + *index += bytes_to_pad; + } + } + RTC_DCHECK_EQ(index_end, *index); + return true; +} + +bool Bye::WithCsrc(uint32_t csrc) { + if (csrcs_.size() >= kMaxNumberOfCsrcs) { + LOG(LS_WARNING) << "Max CSRC size reached."; + return false; + } + csrcs_.push_back(csrc); + return true; +} + +void Bye::WithReason(const std::string& reason) { + RTC_DCHECK_LE(reason.size(), 0xffu); + reason_ = reason; +} + +size_t Bye::BlockLength() const { + size_t src_count = (1 + csrcs_.size()); + size_t reason_size_in_32bits = reason_.empty() ? 0 : (reason_.size() / 4 + 1); + return kHeaderLength + 4 * (src_count + reason_size_in_32bits); +} + +} // namespace rtcp +} // namespace webrtc diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h new file mode 100644 index 0000000000..6b4a181330 --- /dev/null +++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h @@ -0,0 +1,63 @@ +/* + * 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_BYE_H_ +#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_BYE_H_ + +#include <string> +#include <vector> + +#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" +#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" + +namespace webrtc { +namespace rtcp { + +class Bye : public RtcpPacket { + public: + static const uint8_t kPacketType = 203; + + Bye(); + virtual ~Bye() {} + + // 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 WithCsrc(uint32_t csrc); + void WithReason(const std::string& reason); + + uint32_t sender_ssrc() const { return sender_ssrc_; } + const std::vector<uint32_t>& csrcs() const { return csrcs_; } + const std::string& reason() const { return reason_; } + + protected: + bool Create(uint8_t* packet, + size_t* index, + size_t max_length, + RtcpPacket::PacketReadyCallback* callback) const override; + + private: + static const int kMaxNumberOfCsrcs = 0x1f - 1; // First item is sender SSRC. + + size_t BlockLength() const override; + + uint32_t sender_ssrc_; + std::vector<uint32_t> csrcs_; + std::string reason_; + + RTC_DISALLOW_COPY_AND_ASSIGN(Bye); +}; + +} // namespace rtcp +} // namespace webrtc +#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_BYE_H_ diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye_unittest.cc new file mode 100644 index 0000000000..0728ed9552 --- /dev/null +++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye_unittest.cc @@ -0,0 +1,173 @@ +/* + * 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/bye.h" + +#include "testing/gmock/include/gmock/gmock.h" +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" + +using ::testing::ElementsAre; + +using webrtc::rtcp::Bye; +using webrtc::rtcp::RawPacket; +using webrtc::RTCPUtility::RtcpCommonHeader; +using webrtc::RTCPUtility::RtcpParseCommonHeader; + +namespace webrtc { +namespace { + +const uint32_t kSenderSsrc = 0x12345678; +const uint32_t kCsrc1 = 0x22232425; +const uint32_t kCsrc2 = 0x33343536; + +class RtcpPacketByeTest : public ::testing::Test { + protected: + void BuildPacket() { packet = bye.Build().Pass(); } + void ParsePacket() { + RtcpCommonHeader header; + EXPECT_TRUE( + RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header)); + // Check that there is exactly one RTCP packet in the buffer. + EXPECT_EQ(header.BlockSize(), packet->Length()); + EXPECT_TRUE(parsed_bye.Parse( + header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes)); + } + + Bye bye; + rtc::scoped_ptr<RawPacket> packet; + Bye parsed_bye; +}; + +TEST_F(RtcpPacketByeTest, Bye) { + bye.From(kSenderSsrc); + + BuildPacket(); + ParsePacket(); + + EXPECT_EQ(kSenderSsrc, parsed_bye.sender_ssrc()); + EXPECT_TRUE(parsed_bye.csrcs().empty()); + EXPECT_TRUE(parsed_bye.reason().empty()); +} + +TEST_F(RtcpPacketByeTest, WithCsrcs) { + bye.From(kSenderSsrc); + EXPECT_TRUE(bye.WithCsrc(kCsrc1)); + EXPECT_TRUE(bye.WithCsrc(kCsrc2)); + EXPECT_TRUE(bye.reason().empty()); + + BuildPacket(); + EXPECT_EQ(16u, packet->Length()); // Header: 4, 3xSRCs: 12, Reason: 0. + + ParsePacket(); + + EXPECT_EQ(kSenderSsrc, parsed_bye.sender_ssrc()); + EXPECT_THAT(parsed_bye.csrcs(), ElementsAre(kCsrc1, kCsrc2)); + EXPECT_TRUE(parsed_bye.reason().empty()); +} + +TEST_F(RtcpPacketByeTest, WithCsrcsAndReason) { + const std::string kReason = "Some Reason"; + + bye.From(kSenderSsrc); + EXPECT_TRUE(bye.WithCsrc(kCsrc1)); + EXPECT_TRUE(bye.WithCsrc(kCsrc2)); + bye.WithReason(kReason); + + BuildPacket(); + EXPECT_EQ(28u, packet->Length()); // Header: 4, 3xSRCs: 12, Reason: 12. + + ParsePacket(); + + EXPECT_EQ(kSenderSsrc, parsed_bye.sender_ssrc()); + EXPECT_THAT(parsed_bye.csrcs(), ElementsAre(kCsrc1, kCsrc2)); + EXPECT_EQ(kReason, parsed_bye.reason()); +} + +TEST_F(RtcpPacketByeTest, WithTooManyCsrcs) { + bye.From(kSenderSsrc); + const int kMaxCsrcs = (1 << 5) - 2; // 5 bit len, first item is sender SSRC. + for (int i = 0; i < kMaxCsrcs; ++i) { + EXPECT_TRUE(bye.WithCsrc(i)); + } + EXPECT_FALSE(bye.WithCsrc(kMaxCsrcs)); +} + +TEST_F(RtcpPacketByeTest, WithAReason) { + const std::string kReason = "Some Random Reason"; + + bye.From(kSenderSsrc); + bye.WithReason(kReason); + + BuildPacket(); + ParsePacket(); + + EXPECT_EQ(kSenderSsrc, parsed_bye.sender_ssrc()); + EXPECT_TRUE(parsed_bye.csrcs().empty()); + EXPECT_EQ(kReason, parsed_bye.reason()); +} + +TEST_F(RtcpPacketByeTest, WithReasons) { + // Test that packet creation/parsing behave with reasons of different length + // both when it require padding and when it does not. + for (size_t reminder = 0; reminder < 4; ++reminder) { + const std::string kReason(4 + reminder, 'a' + reminder); + bye.From(kSenderSsrc); + bye.WithReason(kReason); + + BuildPacket(); + ParsePacket(); + + EXPECT_EQ(kReason, parsed_bye.reason()); + } +} + +TEST_F(RtcpPacketByeTest, ParseEmptyPacket) { + RtcpCommonHeader header; + header.packet_type = Bye::kPacketType; + header.count_or_format = 0; + header.payload_size_bytes = 0; + uint8_t empty_payload[1]; + + EXPECT_TRUE(parsed_bye.Parse(header, empty_payload + 1)); + EXPECT_EQ(0u, parsed_bye.sender_ssrc()); + EXPECT_TRUE(parsed_bye.csrcs().empty()); + EXPECT_TRUE(parsed_bye.reason().empty()); +} + +TEST_F(RtcpPacketByeTest, ParseFailOnInvalidSrcCount) { + bye.From(kSenderSsrc); + + BuildPacket(); + + RtcpCommonHeader header; + RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header); + header.count_or_format = 2; // Lie there are 2 ssrcs, not one. + + EXPECT_FALSE(parsed_bye.Parse( + header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes)); +} + +TEST_F(RtcpPacketByeTest, ParseFailOnInvalidReasonLength) { + bye.From(kSenderSsrc); + bye.WithReason("18 characters long"); + + BuildPacket(); + + RtcpCommonHeader header; + RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header); + header.payload_size_bytes -= 4; // Payload is usually 32bit aligned. + + EXPECT_FALSE(parsed_bye.Parse( + header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes)); +} + +} // 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 4c1d55e049..3cf3feacd6 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc @@ -15,6 +15,7 @@ #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/test/rtcp_packet_parser.h" using ::testing::ElementsAre; @@ -571,43 +572,6 @@ TEST(RtcpPacketTest, AppendPacketWithOwnAppendedPacket) { EXPECT_EQ(1, parser.fir()->num_packets()); } -TEST(RtcpPacketTest, Bye) { - Bye bye; - bye.From(kSenderSsrc); - - rtc::scoped_ptr<RawPacket> packet(bye.Build()); - RtcpPacketParser parser; - parser.Parse(packet->Buffer(), packet->Length()); - EXPECT_EQ(1, parser.bye()->num_packets()); - EXPECT_EQ(kSenderSsrc, parser.bye()->Ssrc()); -} - -TEST(RtcpPacketTest, ByeWithCsrcs) { - Fir fir; - Bye bye; - bye.From(kSenderSsrc); - EXPECT_TRUE(bye.WithCsrc(0x22222222)); - EXPECT_TRUE(bye.WithCsrc(0x33333333)); - bye.Append(&fir); - - rtc::scoped_ptr<RawPacket> packet(bye.Build()); - RtcpPacketParser parser; - parser.Parse(packet->Buffer(), packet->Length()); - EXPECT_EQ(1, parser.bye()->num_packets()); - EXPECT_EQ(kSenderSsrc, parser.bye()->Ssrc()); - EXPECT_EQ(1, parser.fir()->num_packets()); -} - -TEST(RtcpPacketTest, ByeWithTooManyCsrcs) { - Bye bye; - bye.From(kSenderSsrc); - const int kMaxCsrcs = (1 << 5) - 2; // 5 bit len, first item is sender SSRC. - for (int i = 0; i < kMaxCsrcs; ++i) { - EXPECT_TRUE(bye.WithCsrc(i)); - } - EXPECT_FALSE(bye.WithCsrc(kMaxCsrcs)); -} - TEST(RtcpPacketTest, BuildWithInputBuffer) { Fir fir; ReportBlock rb; diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc index 53c31776ae..5b0e656e32 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc @@ -26,6 +26,7 @@ #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.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/extended_jitter_report.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc index ffab90ebe7..99292994c1 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc +++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc @@ -23,6 +23,7 @@ #include "webrtc/common_types.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.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/transport_feedback.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" |