2ec560606b
This also includes: - Creating new packetizer and depacketizer interfaces. - Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition. - Created a Create() factory method for packetizers and depacketizers. R=niklas.enbom@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21009004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d
55 lines
2.0 KiB
C++
55 lines
2.0 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 "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, size_t max_payload_len);
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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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};
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class RtpDepacketizer {
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public:
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static RtpDepacketizer* Create(RtpVideoCodecTypes type,
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RtpData* const callback);
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virtual ~RtpDepacketizer() {}
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virtual bool Parse(WebRtcRTPHeader* rtp_header,
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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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