ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
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/include/module_common_types.h"
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#include "webrtc/modules/rtp_rtcp/include/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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