346094cb01
When frames are fragmented inte multiple RTP packets in order to not exceed a maximum packet size, the header overhead calculation must take into account that FEC redundancy packets may use more than the 12 bytes of the basic RTP header. For example, a csrc list or extension headers may be present. BUG=2899 R=phoglund@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8769005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5562 4adac7df-926f-26a2-2b94-8c16560cd09d
66 lines
1.8 KiB
C++
66 lines
1.8 KiB
C++
/*
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* Copyright (c) 2012 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_FEC_TEST_HELPER_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
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namespace webrtc {
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enum {
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kFecPayloadType = 96
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};
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enum {
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kRedPayloadType = 97
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};
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enum {
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kVp8PayloadType = 120
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};
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typedef ForwardErrorCorrection::Packet Packet;
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struct RtpPacket : public Packet {
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WebRtcRTPHeader header;
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};
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class FrameGenerator {
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public:
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FrameGenerator();
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void NewFrame(int num_packets);
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uint16_t NextSeqNum();
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RtpPacket* NextPacket(int offset, size_t length);
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// Creates a new RtpPacket with the RED header added to the packet.
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RtpPacket* BuildMediaRedPacket(const RtpPacket* packet);
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// Creates a new RtpPacket with FEC payload and red header. Does this by
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// creating a new fake media RtpPacket, clears the marker bit and adds a RED
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// header. Finally replaces the payload with the content of |packet->data|.
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RtpPacket* BuildFecRedPacket(const Packet* packet);
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void SetRedHeader(Packet* red_packet, uint8_t payload_type,
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int header_length) const;
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private:
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static void BuildRtpHeader(uint8_t* data, const RTPHeader* header);
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int num_packets_;
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uint16_t seq_num_;
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uint32_t timestamp_;
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};
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}
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_
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