d42a3adf42
It' bit ugly that RtpDepacketizer::ParsedPayload partially defines WebRtcRTPHeader, and then sent to Parse() function for internal change. To make it clearer, the CL gets rid of using partially-defined WebRtcRTPHeader. BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28919004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7660 4adac7df-926f-26a2-2b94-8c16560cd09d
73 lines
2.5 KiB
C++
73 lines
2.5 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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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
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#include <string>
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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namespace webrtc {
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class RtpPacketizer {
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public:
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static RtpPacketizer* Create(RtpVideoCodecTypes type,
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size_t max_payload_len,
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const RTPVideoTypeHeader* rtp_type_header,
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FrameType frame_type);
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virtual ~RtpPacketizer() {}
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virtual void SetPayloadData(const uint8_t* payload_data,
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size_t payload_size,
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const RTPFragmentationHeader* fragmentation) = 0;
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// Get the next payload with payload header.
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// buffer is a pointer to where the output will be written.
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// bytes_to_send is an output variable that will contain number of bytes
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// written to buffer. The parameter last_packet is true for the last packet of
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// the frame, false otherwise (i.e., call the function again to get the
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// next packet).
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// Returns true on success or false if there was no payload to packetize.
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virtual bool NextPacket(uint8_t* buffer,
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size_t* bytes_to_send,
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bool* last_packet) = 0;
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virtual ProtectionType GetProtectionType() = 0;
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virtual StorageType GetStorageType(uint32_t retransmission_settings) = 0;
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virtual std::string ToString() = 0;
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};
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class RtpDepacketizer {
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public:
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struct ParsedPayload {
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const uint8_t* payload;
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size_t payload_length;
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FrameType frame_type;
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RTPTypeHeader type;
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};
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static RtpDepacketizer* Create(RtpVideoCodecTypes type);
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virtual ~RtpDepacketizer() {}
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// Parses the RTP payload, parsed result will be saved in |parsed_payload|.
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virtual bool Parse(ParsedPayload* parsed_payload,
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const uint8_t* payload_data,
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size_t payload_data_length) = 0;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
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