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988cc0870b
This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
100 lines
3.6 KiB
C++
100 lines
3.6 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/rtp_rtcp/source/rtcp_packet/app.h"
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#include <string.h>
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#include <cstdint>
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#include "modules/rtp_rtcp/source/byte_io.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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namespace webrtc {
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namespace rtcp {
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constexpr uint8_t App::kPacketType;
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constexpr size_t App::kMaxDataSize;
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// Application-Defined packet (APP) (RFC 3550).
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//
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// 0 1 2 3
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// 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
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// |V=2|P| subtype | PT=APP=204 | length |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// 0 | SSRC/CSRC |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// 4 | name (ASCII) |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// 8 | application-dependent data ...
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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App::App() : sub_type_(0), ssrc_(0), name_(0) {}
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App::~App() = default;
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bool App::Parse(const CommonHeader& packet) {
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RTC_DCHECK_EQ(packet.type(), kPacketType);
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if (packet.payload_size_bytes() < kAppBaseLength) {
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RTC_LOG(LS_WARNING) << "Packet is too small to be a valid APP packet";
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return false;
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}
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if (packet.payload_size_bytes() % 4 != 0) {
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RTC_LOG(LS_WARNING)
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<< "Packet payload must be 32 bits aligned to make a valid APP packet";
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return false;
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}
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sub_type_ = packet.fmt();
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ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&packet.payload()[0]);
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name_ = ByteReader<uint32_t>::ReadBigEndian(&packet.payload()[4]);
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data_.SetData(packet.payload() + kAppBaseLength,
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packet.payload_size_bytes() - kAppBaseLength);
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return true;
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}
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void App::SetSubType(uint8_t subtype) {
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RTC_DCHECK_LE(subtype, 0x1f);
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sub_type_ = subtype;
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}
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void App::SetData(const uint8_t* data, size_t data_length) {
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RTC_DCHECK(data);
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RTC_DCHECK_EQ(data_length % 4, 0) << "Data must be 32 bits aligned.";
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RTC_DCHECK_LE(data_length, kMaxDataSize)
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<< "App data size " << data_length << " exceed maximum of "
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<< kMaxDataSize << " bytes.";
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data_.SetData(data, data_length);
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}
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size_t App::BlockLength() const {
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return kHeaderLength + kAppBaseLength + data_.size();
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}
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bool App::Create(uint8_t* packet,
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size_t* index,
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size_t max_length,
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PacketReadyCallback callback) const {
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while (*index + BlockLength() > max_length) {
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if (!OnBufferFull(packet, index, callback))
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return false;
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}
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const size_t index_end = *index + BlockLength();
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CreateHeader(sub_type_, kPacketType, HeaderLength(), packet, index);
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ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 0], ssrc_);
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ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 4], name_);
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memcpy(&packet[*index + 8], data_.data(), data_.size());
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*index += (8 + data_.size());
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RTC_DCHECK_EQ(index_end, *index);
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return true;
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}
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} // namespace rtcp
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} // namespace webrtc
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