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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.cc

Issue 1544983002: [rtp_rtcp] rtcp::SenderReport moved into own file and got Parse function (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 4 years, 11 months ago
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1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
3 *
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
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
12
13 #include "webrtc/base/checks.h"
14 #include "webrtc/base/logging.h"
15 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
16
17 using webrtc::RTCPUtility::RtcpCommonHeader;
18
19 namespace webrtc {
20 namespace rtcp {
21 // Sender report (SR) (RFC 3550).
22 // 0 1 2 3
23 // 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
24 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
25 // |V=2|P| RC | PT=SR=200 | length |
26 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
27 // 0 | SSRC of sender |
28 // +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
29 // 4 | NTP timestamp, most significant word |
30 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
31 // 8 | NTP timestamp, least significant word |
32 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
33 // 12 | RTP timestamp |
34 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
35 // 16 | sender's packet count |
36 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
37 // 20 | sender's octet count |
38 // 24 +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
39
40 SenderReport::SenderReport()
41 : sender_ssrc_(0),
42 rtp_timestamp_(0),
43 sender_packet_count_(0),
44 sender_octet_count_(0) {}
45
46 bool SenderReport::Parse(const RtcpCommonHeader& header,
47 const uint8_t* payload) {
48 RTC_DCHECK(header.packet_type == kPacketType);
49
50 const uint8_t report_block_count = header.count_or_format;
51 if (header.payload_size_bytes <
52 kSenderBaseLength + report_block_count * ReportBlock::kLength) {
53 LOG(LS_WARNING) << "Packet is too small to contain all the data.";
54 return false;
55 }
56 // Read SenderReport header.
57 sender_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&payload[0]);
58 uint32_t secs = ByteReader<uint32_t>::ReadBigEndian(&payload[4]);
59 uint32_t frac = ByteReader<uint32_t>::ReadBigEndian(&payload[8]);
60 ntp_.Set(secs, frac);
61 rtp_timestamp_ = ByteReader<uint32_t>::ReadBigEndian(&payload[12]);
62 sender_packet_count_ = ByteReader<uint32_t>::ReadBigEndian(&payload[16]);
63 sender_octet_count_ = ByteReader<uint32_t>::ReadBigEndian(&payload[20]);
64 report_blocks_.resize(report_block_count);
65 const uint8_t* next_block = payload + kSenderBaseLength;
66 for (ReportBlock& block : report_blocks_) {
67 bool block_parsed = block.Parse(next_block, ReportBlock::kLength);
68 RTC_DCHECK(block_parsed);
69 next_block += ReportBlock::kLength;
70 }
71 // Double check we didn't read beyond provided buffer.
72 RTC_DCHECK_LE(next_block, payload + header.payload_size_bytes);
73 return true;
74 }
75
76 bool SenderReport::Create(uint8_t* packet,
77 size_t* index,
78 size_t max_length,
79 RtcpPacket::PacketReadyCallback* callback) const {
80 while (*index + BlockLength() > max_length) {
81 if (!OnBufferFull(packet, index, callback))
82 return false;
83 }
84 const size_t index_end = *index + BlockLength();
85
86 CreateHeader(report_blocks_.size(), kPacketType, HeaderLength(), packet,
87 index);
88 // Write SenderReport header.
89 ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 0], sender_ssrc_);
90 ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 4], ntp_.seconds());
91 ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 8], ntp_.fractions());
92 ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 12], rtp_timestamp_);
93 ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 16],
94 sender_packet_count_);
95 ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 20],
96 sender_octet_count_);
97 *index += kSenderBaseLength;
98 // Write report blocks.
99 for (const ReportBlock& block : report_blocks_) {
100 block.Create(packet + *index);
101 *index += ReportBlock::kLength;
102 }
103 // Ensure bytes written match expected.
104 RTC_DCHECK_EQ(*index, index_end);
105 return true;
106 }
107
108 bool SenderReport::WithReportBlock(const ReportBlock& block) {
109 if (report_blocks_.size() >= kMaxNumberOfReportBlocks) {
110 LOG(LS_WARNING) << "Max report blocks reached.";
111 return false;
112 }
113 report_blocks_.push_back(block);
114 return true;
115 }
116
117 } // namespace rtcp
118 } // namespace webrtc
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