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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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722 return true; | 722 return true; |
723 } | 723 } |
724 | 724 |
725 size_t RtcpPacket::HeaderLength() const { | 725 size_t RtcpPacket::HeaderLength() const { |
726 size_t length_in_bytes = BlockLength(); | 726 size_t length_in_bytes = BlockLength(); |
727 // Length in 32-bit words minus 1. | 727 // Length in 32-bit words minus 1. |
728 assert(length_in_bytes > 0); | 728 assert(length_in_bytes > 0); |
729 return ((length_in_bytes + 3) / 4) - 1; | 729 return ((length_in_bytes + 3) / 4) - 1; |
730 } | 730 } |
731 | 731 |
732 RtcpPacket::CommonHeader::CommonHeader() | |
733 : count_or_format(0), payload_type(0), payload_size_bytes(0) {} | |
732 // From RFC 3550, RTP: A Transport Protocol for Real-Time Applications. | 734 // From RFC 3550, RTP: A Transport Protocol for Real-Time Applications. |
733 // | 735 // |
734 // RTP header format. | 736 // RTP header format. |
735 // 0 1 2 3 | 737 // 0 1 2 3 |
736 // 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 | 738 // 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 |
737 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | 739 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
738 // |V=2|P| RC/FMT | PT | length | | 740 // |V=2|P| RC/FMT | PT | length | |
739 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | 741 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
740 | 742 |
741 void RtcpPacket::CreateHeader( | 743 void RtcpPacket::CreateHeader( |
742 uint8_t count_or_format, // Depends on packet type. | 744 uint8_t count_or_format, // Depends on packet type. |
743 uint8_t packet_type, | 745 uint8_t packet_type, |
744 size_t length, | 746 size_t length, |
745 uint8_t* buffer, | 747 uint8_t* buffer, |
746 size_t* pos) const { | 748 size_t* pos) { |
747 assert(length <= 0xffff); | 749 assert(length <= 0xffff); |
748 const uint8_t kVersion = 2; | 750 const uint8_t kVersion = 2; |
749 AssignUWord8(buffer, pos, (kVersion << 6) + count_or_format); | 751 AssignUWord8(buffer, pos, (kVersion << 6) + count_or_format); |
750 AssignUWord8(buffer, pos, packet_type); | 752 AssignUWord8(buffer, pos, packet_type); |
751 AssignUWord16(buffer, pos, length); | 753 AssignUWord16(buffer, pos, length); |
752 } | 754 } |
753 | 755 |
756 bool RtcpPacket::ParseHeader(const uint8_t* packet, | |
757 size_t index, | |
758 size_t max_length, | |
759 CommonHeader* header) { | |
åsapersson
2015/09/03 08:57:20
Looks like RTCPUtility::RTCPParseCommonHeader can
sprang_webrtc
2015/09/04 14:12:50
I've merged this into RTCPUtility.
| |
760 DCHECK(header != nullptr); | |
761 size_t length = max_length - index; | |
762 const size_t kHeaderLength = 4; | |
763 if (length < kHeaderLength) { | |
764 LOG(LS_WARNING) << "Too little data (" << length << " byte" | |
765 << (length != 1 ? "s" : "") | |
766 << ") remaining in buffer to parse RTCP header (4 bytes)."; | |
767 return false; | |
768 } | |
769 | |
770 const uint8_t kRtcpVersion = 2; | |
771 uint8_t version = packet[index] >> 6; | |
772 if (version != kRtcpVersion) { | |
773 LOG(LS_WARNING) << "Invalid RTCP header: Version must be " << kRtcpVersion | |
774 << " but was " << version; | |
775 return false; | |
776 } | |
777 | |
778 size_t packet_size_words = | |
779 ByteReader<uint16_t>::ReadBigEndian(&packet[index + 2]) + 1; | |
780 if (length < packet_size_words * 4) { | |
781 LOG(LS_WARNING) << "Buffer too small (" << length | |
782 << " bytes) to fit a FeedbackPacket of " | |
783 << packet_size_words << " 32bit words."; | |
784 return false; | |
785 } | |
786 | |
787 bool has_padding = (packet[index] & 0x20) != 0; | |
788 size_t payload_size = packet_size_words * 4; | |
789 if (has_padding) { | |
790 uint8_t padding_bytes = packet[index + payload_size - 1]; | |
791 if (kHeaderLength + padding_bytes > payload_size) { | |
792 LOG(LS_WARNING) << "Invalid RTCP header: Too many padding bytes (" | |
793 << padding_bytes << ") for a packet size of " | |
794 << payload_size << "bytes."; | |
795 return false; | |
796 } | |
797 payload_size -= padding_bytes; | |
798 } | |
799 payload_size -= kHeaderLength; | |
800 | |
801 header->payload_size_bytes = payload_size; | |
802 header->count_or_format = packet[index] & 0x1F; | |
803 header->payload_type = packet[index + 1]; | |
804 | |
805 return true; | |
806 } | |
807 | |
754 bool Empty::Create(uint8_t* packet, | 808 bool Empty::Create(uint8_t* packet, |
755 size_t* index, | 809 size_t* index, |
756 size_t max_length, | 810 size_t max_length, |
757 RtcpPacket::PacketReadyCallback* callback) const { | 811 RtcpPacket::PacketReadyCallback* callback) const { |
758 return true; | 812 return true; |
759 } | 813 } |
760 | 814 |
761 size_t Empty::BlockLength() const { | 815 size_t Empty::BlockLength() const { |
762 return 0; | 816 return 0; |
763 } | 817 } |
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1220 return length_; | 1274 return length_; |
1221 } | 1275 } |
1222 | 1276 |
1223 void RawPacket::SetLength(size_t length) { | 1277 void RawPacket::SetLength(size_t length) { |
1224 assert(length <= buffer_length_); | 1278 assert(length <= buffer_length_); |
1225 length_ = length; | 1279 length_ = length; |
1226 } | 1280 } |
1227 | 1281 |
1228 } // namespace rtcp | 1282 } // namespace rtcp |
1229 } // namespace webrtc | 1283 } // namespace webrtc |
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