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}
45 lines
1.6 KiB
C++
45 lines
1.6 KiB
C++
/*
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* Copyright (c) 2013 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_INCLUDE_RTP_HEADER_PARSER_H_
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#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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struct RTPHeader;
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class RtpHeaderParser {
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public:
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static RtpHeaderParser* Create();
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virtual ~RtpHeaderParser() {}
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// Returns true if the packet is an RTCP packet, false otherwise.
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static bool IsRtcp(const uint8_t* packet, size_t length);
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// Parses the packet and stores the parsed packet in |header|. Returns true on
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// success, false otherwise.
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// This method is thread-safe in the sense that it can parse multiple packets
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// at once.
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virtual bool Parse(const uint8_t* packet,
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size_t length,
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RTPHeader* header) const = 0;
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// Registers an RTP header extension and binds it to |id|.
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virtual bool RegisterRtpHeaderExtension(RTPExtensionType type,
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uint8_t id) = 0;
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// De-registers an RTP header extension.
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virtual bool DeregisterRtpHeaderExtension(RTPExtensionType type) = 0;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
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