Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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modules/rtp_rtcp/source/rtcp_packet/bye.cc
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137
modules/rtp_rtcp/source/rtcp_packet/bye.cc
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/*
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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 "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
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#include <utility>
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#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
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#include "webrtc/rtc_base/checks.h"
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#include "webrtc/rtc_base/logging.h"
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namespace webrtc {
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namespace rtcp {
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constexpr uint8_t Bye::kPacketType;
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// Bye packet (BYE) (RFC 3550).
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//
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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| SC | PT=BYE=203 | length |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// | SSRC/CSRC |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// : ... :
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// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
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// (opt) | length | reason for leaving ...
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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Bye::Bye() : sender_ssrc_(0) {}
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Bye::~Bye() = default;
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bool Bye::Parse(const CommonHeader& packet) {
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RTC_DCHECK_EQ(packet.type(), kPacketType);
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const uint8_t src_count = packet.count();
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// Validate packet.
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if (packet.payload_size_bytes() < 4u * src_count) {
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LOG(LS_WARNING)
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<< "Packet is too small to contain CSRCs it promise to have.";
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return false;
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}
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const uint8_t* const payload = packet.payload();
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bool has_reason = packet.payload_size_bytes() > 4u * src_count;
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uint8_t reason_length = 0;
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if (has_reason) {
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reason_length = payload[4u * src_count];
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if (packet.payload_size_bytes() - 4u * src_count < 1u + reason_length) {
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LOG(LS_WARNING) << "Invalid reason length: " << reason_length;
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return false;
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}
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}
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// Once sure packet is valid, copy values.
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if (src_count == 0) { // A count value of zero is valid, but useless.
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sender_ssrc_ = 0;
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csrcs_.clear();
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} else {
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sender_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(payload);
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csrcs_.resize(src_count - 1);
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for (size_t i = 1; i < src_count; ++i)
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csrcs_[i - 1] = ByteReader<uint32_t>::ReadBigEndian(&payload[4 * i]);
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}
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if (has_reason) {
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reason_.assign(reinterpret_cast<const char*>(&payload[4u * src_count + 1]),
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reason_length);
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} else {
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reason_.clear();
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}
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return true;
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}
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bool Bye::Create(uint8_t* packet,
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size_t* index,
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size_t max_length,
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RtcpPacket::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(1 + csrcs_.size(), kPacketType, HeaderLength(), packet, index);
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// Store srcs of the leaving clients.
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ByteWriter<uint32_t>::WriteBigEndian(&packet[*index], sender_ssrc_);
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*index += sizeof(uint32_t);
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for (uint32_t csrc : csrcs_) {
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ByteWriter<uint32_t>::WriteBigEndian(&packet[*index], csrc);
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*index += sizeof(uint32_t);
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}
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// Store the reason to leave.
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if (!reason_.empty()) {
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uint8_t reason_length = reason_.size();
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packet[(*index)++] = reason_length;
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memcpy(&packet[*index], reason_.data(), reason_length);
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*index += reason_length;
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// Add padding bytes if needed.
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size_t bytes_to_pad = index_end - *index;
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RTC_DCHECK_LE(bytes_to_pad, 3);
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if (bytes_to_pad > 0) {
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memset(&packet[*index], 0, bytes_to_pad);
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*index += bytes_to_pad;
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}
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}
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RTC_DCHECK_EQ(index_end, *index);
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return true;
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}
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bool Bye::SetCsrcs(std::vector<uint32_t> csrcs) {
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if (csrcs.size() > kMaxNumberOfCsrcs) {
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LOG(LS_WARNING) << "Too many CSRCs for Bye packet.";
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return false;
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}
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csrcs_ = std::move(csrcs);
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return true;
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}
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void Bye::SetReason(std::string reason) {
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RTC_DCHECK_LE(reason.size(), 0xffu);
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reason_ = std::move(reason);
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}
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size_t Bye::BlockLength() const {
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size_t src_count = (1 + csrcs_.size());
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size_t reason_size_in_32bits = reason_.empty() ? 0 : (reason_.size() / 4 + 1);
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return kHeaderLength + 4 * (src_count + reason_size_in_32bits);
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}
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} // namespace rtcp
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} // namespace webrtc
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