20e77c7b8a
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 Review-Url: https://codereview.webrtc.org/2348623003 Cr-Commit-Position: refs/heads/master@{#14393}
108 lines
3.7 KiB
C++
108 lines
3.7 KiB
C++
/*
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* Copyright (c) 2014 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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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
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#include "webrtc/base/buffer.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
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namespace webrtc {
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namespace rtcp {
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// Class for building RTCP packets.
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//
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// Example:
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// ReportBlock report_block;
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// report_block.SetMediaSsrc(234);
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// report_block.SetFractionLost(10);
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//
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// ReceiverReport rr;
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// rr.SetSenderSsrc(123);
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// rr.AddReportBlock(report_block);
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//
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// Fir fir;
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// fir.SetSenderSsrc(123);
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// fir.AddRequestTo(234, 56);
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//
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// size_t length = 0; // Builds an intra frame request
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// uint8_t packet[kPacketSize]; // with sequence number 56.
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// fir.Build(packet, &length, kPacketSize);
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//
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// rtc::Buffer packet = fir.Build(); // Returns a RawPacket holding
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// // the built rtcp packet.
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//
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// CompoundPacket compound; // Builds a compound RTCP packet with
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// compound.Append(&rr); // a receiver report, report block
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// compound.Append(&fir); // and fir message.
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// rtc::Buffer packet = compound.Build();
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class RtcpPacket {
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public:
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virtual ~RtcpPacket() {}
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// Callback used to signal that an RTCP packet is ready. Note that this may
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// not contain all data in this RtcpPacket; if a packet cannot fit in
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// max_length bytes, it will be fragmented and multiple calls to this
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// callback will be made.
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class PacketReadyCallback {
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public:
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PacketReadyCallback() {}
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virtual ~PacketReadyCallback() {}
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virtual void OnPacketReady(uint8_t* data, size_t length) = 0;
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};
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// Convenience method mostly used for test. Max length of IP_PACKET_SIZE is
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// used, will cause assertion error if fragmentation occurs.
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rtc::Buffer Build() const;
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// Returns true if call to Create succeeded. A buffer of size
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// IP_PACKET_SIZE will be allocated and reused between calls to callback.
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bool Build(PacketReadyCallback* callback) const;
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// Returns true if call to Create succeeded. Provided buffer reference
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// will be used for all calls to callback.
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bool BuildExternalBuffer(uint8_t* buffer,
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size_t max_length,
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PacketReadyCallback* callback) const;
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// Size of this packet in bytes (including headers).
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virtual size_t BlockLength() const = 0;
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// Creates packet in the given buffer at the given position.
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// Calls PacketReadyCallback::OnPacketReady if remaining buffer is too small
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// and assume buffer can be reused after OnPacketReady returns.
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virtual bool 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 = 0;
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protected:
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RtcpPacket() {}
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static void CreateHeader(uint8_t count_or_format,
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uint8_t packet_type,
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size_t block_length, // Size in 32bit words - 1.
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uint8_t* buffer,
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size_t* pos);
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bool OnBufferFull(uint8_t* packet,
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size_t* index,
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RtcpPacket::PacketReadyCallback* callback) const;
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size_t HeaderLength() const;
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static const size_t kHeaderLength = 4;
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};
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} // namespace rtcp
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
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