| /* |
| * Copyright (c) 2014 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. |
| * |
| */ |
| #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_ |
| #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_ |
| |
| #include "webrtc/rtc_base/basictypes.h" |
| #include "webrtc/rtc_base/buffer.h" |
| |
| namespace webrtc { |
| namespace rtcp { |
| // Class for building RTCP packets. |
| // |
| // Example: |
| // ReportBlock report_block; |
| // report_block.SetMediaSsrc(234); |
| // report_block.SetFractionLost(10); |
| // |
| // ReceiverReport rr; |
| // rr.SetSenderSsrc(123); |
| // rr.AddReportBlock(report_block); |
| // |
| // Fir fir; |
| // fir.SetSenderSsrc(123); |
| // fir.AddRequestTo(234, 56); |
| // |
| // size_t length = 0; // Builds an intra frame request |
| // uint8_t packet[kPacketSize]; // with sequence number 56. |
| // fir.Build(packet, &length, kPacketSize); |
| // |
| // rtc::Buffer packet = fir.Build(); // Returns a RawPacket holding |
| // // the built rtcp packet. |
| // |
| // CompoundPacket compound; // Builds a compound RTCP packet with |
| // compound.Append(&rr); // a receiver report, report block |
| // compound.Append(&fir); // and fir message. |
| // rtc::Buffer packet = compound.Build(); |
| |
| class RtcpPacket { |
| public: |
| // Callback used to signal that an RTCP packet is ready. Note that this may |
| // not contain all data in this RtcpPacket; if a packet cannot fit in |
| // max_length bytes, it will be fragmented and multiple calls to this |
| // callback will be made. |
| class PacketReadyCallback { |
| public: |
| virtual void OnPacketReady(uint8_t* data, size_t length) = 0; |
| |
| protected: |
| PacketReadyCallback() {} |
| virtual ~PacketReadyCallback() {} |
| }; |
| |
| virtual ~RtcpPacket() {} |
| |
| // Convenience method mostly used for test. Creates packet without |
| // fragmentation using BlockLength() to allocate big enough buffer. |
| rtc::Buffer Build() const; |
| |
| // Returns true if call to Create succeeded. Provided buffer reference |
| // will be used for all calls to callback. |
| bool BuildExternalBuffer(uint8_t* buffer, |
| size_t max_length, |
| PacketReadyCallback* callback) const; |
| |
| // Size of this packet in bytes (including headers). |
| virtual size_t BlockLength() const = 0; |
| |
| // Creates packet in the given buffer at the given position. |
| // Calls PacketReadyCallback::OnPacketReady if remaining buffer is too small |
| // and assume buffer can be reused after OnPacketReady returns. |
| virtual bool Create(uint8_t* packet, |
| size_t* index, |
| size_t max_length, |
| PacketReadyCallback* callback) const = 0; |
| |
| protected: |
| // Size of the rtcp common header. |
| static constexpr size_t kHeaderLength = 4; |
| RtcpPacket() {} |
| |
| static void CreateHeader(uint8_t count_or_format, |
| uint8_t packet_type, |
| size_t block_length, // Payload size in 32bit words. |
| uint8_t* buffer, |
| size_t* pos); |
| |
| bool OnBufferFull(uint8_t* packet, |
| size_t* index, |
| PacketReadyCallback* callback) const; |
| // Size of the rtcp packet as written in header. |
| size_t HeaderLength() const; |
| }; |
| } // namespace rtcp |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_ |