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Moving src/webrtc into src/.

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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}
This commit is contained in:
Mirko Bonadei
2017-09-15 06:15:48 +02:00
committed by Commit Bot
parent 6674846b4a
commit bb547203bf
4576 changed files with 1092 additions and 1196 deletions
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97
modules/rtp_rtcp/source/rtcp_packet/app.cc Normal file
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/*
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/logging.h"
namespace webrtc {
namespace rtcp {
constexpr uint8_t App::kPacketType;
constexpr size_t App::kMaxDataSize;
// Application-Defined packet (APP) (RFC 3550).
//
// 0 1 2 3
// 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
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// |V=2|P| subtype | PT=APP=204 | length |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// 0 | SSRC/CSRC |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// 4 | name (ASCII) |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// 8 | application-dependent data ...
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
App::App() : sub_type_(0), ssrc_(0), name_(0) {}
App::~App() = default;
bool App::Parse(const CommonHeader& packet) {
RTC_DCHECK_EQ(packet.type(), kPacketType);
if (packet.payload_size_bytes() < kAppBaseLength) {
LOG(LS_WARNING) << "Packet is too small to be a valid APP packet";
return false;
}
if (packet.payload_size_bytes() % 4 != 0) {
LOG(LS_WARNING)
<< "Packet payload must be 32 bits aligned to make a valid APP packet";
return false;
}
sub_type_ = packet.fmt();
ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&packet.payload()[0]);
name_ = ByteReader<uint32_t>::ReadBigEndian(&packet.payload()[4]);
data_.SetData(packet.payload() + kAppBaseLength,
packet.payload_size_bytes() - kAppBaseLength);
return true;
}
void App::SetSubType(uint8_t subtype) {
RTC_DCHECK_LE(subtype, 0x1f);
sub_type_ = subtype;
}
void App::SetData(const uint8_t* data, size_t data_length) {
RTC_DCHECK(data);
RTC_DCHECK_EQ(data_length % 4, 0) << "Data must be 32 bits aligned.";
RTC_DCHECK_LE(data_length, kMaxDataSize) << "App data size " << data_length
<< " exceed maximum of "
<< kMaxDataSize << " bytes.";
data_.SetData(data, data_length);
}
size_t App::BlockLength() const {
return kHeaderLength + kAppBaseLength + data_.size();
}
bool App::Create(uint8_t* packet,
size_t* index,
size_t max_length,
RtcpPacket::PacketReadyCallback* callback) const {
while (*index + BlockLength() > max_length) {
if (!OnBufferFull(packet, index, callback))
return false;
}
const size_t index_end = *index + BlockLength();
CreateHeader(sub_type_, kPacketType, HeaderLength(), packet, index);
ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 0], ssrc_);
ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 4], name_);
memcpy(&packet[*index + 8], data_.data(), data_.size());
*index += (8 + data_.size());
RTC_DCHECK_EQ(index_end, *index);
return true;
}
} // namespace rtcp
} // namespace webrtc