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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}
103 lines
3.6 KiB
C++
103 lines
3.6 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/rtc_base/basictypes.h"
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#include "webrtc/rtc_base/buffer.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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// 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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virtual void OnPacketReady(uint8_t* data, size_t length) = 0;
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protected:
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PacketReadyCallback() {}
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virtual ~PacketReadyCallback() {}
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};
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virtual ~RtcpPacket() {}
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// Convenience method mostly used for test. Creates packet without
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// fragmentation using BlockLength() to allocate big enough buffer.
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rtc::Buffer Build() 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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// Size of the rtcp common header.
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static constexpr size_t kHeaderLength = 4;
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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, // Payload size in 32bit words.
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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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PacketReadyCallback* callback) const;
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// Size of the rtcp packet as written in header.
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size_t HeaderLength() const;
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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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