webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc
kjellander efc6e41866 Revert of Unify rtcp packet setters (patchset #8 id:130001 of https://codereview.webrtc.org/2348623003/ )
Reason for revert:
Breaks compilation of internal downstream project.

Original issue's description:
> Unify rtcp packet setters
> Renamed setters in rtcp classes
> from WithField to SetField
> from WithItem to AddItem or SetItems
> from From to SetSenderSsrc
> from To to SetMediaSsrc
> Some redundant or unsued setters removed.
> Pass-by-const& replaced with pass-by-value when appropriate.
>
> BUG=webrtc:5260
>
> Committed: https://crrev.com/20e77c7b8a9f19942ef3c3c4f1fa3888b2cd54ea
> Cr-Commit-Position: refs/heads/master@{#14393}

TBR=sprang@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5260

Review-Url: https://codereview.webrtc.org/2372713005
Cr-Commit-Position: refs/heads/master@{#14400}
2016-09-27 15:39:39 +00:00

88 lines
3.5 KiB
C++

/*
* 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/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.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 ...
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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::WithSubType(uint8_t subtype) {
RTC_DCHECK_LE(subtype, 0x1f);
sub_type_ = subtype;
}
void App::WithData(const uint8_t* data, size_t data_length) {
RTC_DCHECK(data);
RTC_DCHECK_EQ(data_length % 4, 0u) << "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);
}
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