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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
70 lines
2.4 KiB
C++
70 lines
2.4 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 MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
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#include <string>
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#include "modules/include/module_common_types.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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class RtpPacketToSend;
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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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size_t last_packet_reduction_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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// Returns total number of packets which would be produced by the packetizer.
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virtual size_t SetPayloadData(
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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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// Write payload and set marker bit of the |packet|.
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// Returns true on success, false otherwise.
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virtual bool NextPacket(RtpPacketToSend* packet) = 0;
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virtual std::string ToString() = 0;
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};
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// TODO(sprang): Update the depacketizer to return a std::unqie_ptr with a copy
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// of the parsed payload, rather than just a pointer into the incoming buffer.
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// This way we can move some parsing out from the jitter buffer into here, and
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// the jitter buffer can just store that pointer rather than doing a copy there.
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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 // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
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