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

Issue 1437353003: rtcp::App moved into own file and got Parse function (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 1 month ago
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1 /*
2 * Copyright (c) 2015 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/app.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 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
17
18 using webrtc::RTCPUtility::RtcpCommonHeader;
19
20 namespace webrtc {
21 namespace rtcp {
22
23 // Application-Defined packet (APP) (RFC 3550).
24 //
25 // 0 1 2 3
26 // 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
27 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
28 // |V=2|P| subtype | PT=APP=204 | length |
29 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
30 // 0 | SSRC/CSRC |
31 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
32 // 4 | name (ASCII) |
33 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
34 // 8 | application-dependent data ...
35 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
36 bool App::Parse(const RtcpCommonHeader& header, const uint8_t* payload) {
37 RTC_DCHECK(header.packet_type == kPacketType);
38
39 sub_type_ = header.count_or_format;
40 ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&payload[0]);
41 name_ = ByteReader<uint32_t>::ReadBigEndian(&payload[4]);
42 data_.SetData(&payload[8], header.payload_size_bytes - 8);
43 return true;
44 }
45
46 bool App::WithSubType(uint8_t subtype) {
47 if (subtype > 0x1f) {
åsapersson 2015/11/17 15:31:36 Should we use RTC_DCHECK?
danilchap 2015/11/17 17:12:36 Done.
åsapersson 2015/11/18 10:36:55 Maybe RTC_DCHECK_LE(subtype, 0x1f);
danilchap 2015/11/18 11:15:54 Yes, got confused by RTC_DCHECK_EQ where it is com
48 LOG(LS_WARNING) << "Incorrect subtype " << subtype << " for APP packet.";
49 return false;
50 }
51 sub_type_ = subtype;
52 return true;
53 }
54
55 bool App::WithData(const uint8_t* data, size_t data_length) {
56 RTC_DCHECK(data);
57 if (data_length > kMaxDataSize) {
58 LOG(LS_WARNING) << "App data size << " << data_length
59 << "exceed maximum of " << kMaxDataSize << " bytes.";
60 return false;
61 }
62 if (data_length % 4 != 0) {
åsapersson 2015/11/17 15:31:36 Should we use RTC_DCHECK?
danilchap 2015/11/17 17:12:36 Done.
63 LOG(LS_WARNING) << "Data must be 32 bits aligned.";
64 return false;
65 }
66 data_.SetData(data, data_length);
67 return true;
68 }
69
70 bool App::Create(uint8_t* packet,
71 size_t* index,
72 size_t max_length,
73 RtcpPacket::PacketReadyCallback* callback) const {
74 while (*index + BlockLength() > max_length) {
75 if (!OnBufferFull(packet, index, callback))
76 return false;
77 }
78 const size_t index_end = *index + BlockLength();
79 CreateHeader(sub_type_, kPacketType, HeaderLength(), packet, index);
80
81 ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 0], ssrc_);
82 ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 4], name_);
83 memcpy(&packet[*index + 8], data_.data(), data_.size());
84 *index += (8 + data_.size());
85 RTC_DCHECK_EQ(index_end, *index);
86 return true;
87 }
88
89 } // namespace rtcp
90 } // namespace webrtc
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