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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: merged with master 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
17 using webrtc::RTCPUtility::RtcpCommonHeader;
18
19 namespace webrtc {
20 namespace rtcp {
21
22 // Application-Defined packet (APP) (RFC 3550).
23 //
24 // 0 1 2 3
25 // 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
26 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
27 // |V=2|P| subtype | PT=APP=204 | length |
28 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
29 // 0 | SSRC/CSRC |
30 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
31 // 4 | name (ASCII) |
32 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
33 // 8 | application-dependent data ...
34 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
35 bool App::Parse(const RtcpCommonHeader& header, const uint8_t* payload) {
36 RTC_DCHECK(header.packet_type == kPacketType);
37
38 sub_type_ = header.count_or_format;
39 ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&payload[0]);
40 name_ = ByteReader<uint32_t>::ReadBigEndian(&payload[4]);
41 data_.SetData(&payload[8], header.payload_size_bytes - 8);
42 return true;
43 }
44
45 void App::WithSubType(uint8_t subtype) {
46 RTC_DCHECK_LE(subtype, 0x1f);
47 sub_type_ = subtype;
48 }
49
50 void App::WithData(const uint8_t* data, size_t data_length) {
51 RTC_DCHECK(data);
52 RTC_DCHECK_EQ(0u, data_length % 4) << "Data must be 32 bits aligned.";
53 RTC_DCHECK(data_length <= kMaxDataSize) << "App data size << " << data_length
54 << "exceed maximum of "
55 << kMaxDataSize << " bytes.";
56 data_.SetData(data, data_length);
57 }
58
59 bool App::Create(uint8_t* packet,
60 size_t* index,
61 size_t max_length,
62 RtcpPacket::PacketReadyCallback* callback) const {
63 while (*index + BlockLength() > max_length) {
64 if (!OnBufferFull(packet, index, callback))
65 return false;
66 }
67 const size_t index_end = *index + BlockLength();
68 CreateHeader(sub_type_, kPacketType, HeaderLength(), packet, index);
69
70 ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 0], ssrc_);
71 ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 4], name_);
72 memcpy(&packet[*index + 8], data_.data(), data_.size());
73 *index += (8 + data_.size());
74 RTC_DCHECK_EQ(index_end, *index);
75 return true;
76 }
77
78 } // namespace rtcp
79 } // namespace webrtc
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