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}
60 lines
1.9 KiB
C++
60 lines
1.9 KiB
C++
/*
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* Copyright (c) 2011 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_VIDEO_CODING_PACKET_H_
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#define WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class VCMPacket {
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public:
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VCMPacket();
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VCMPacket(const uint8_t* ptr,
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const size_t size,
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const WebRtcRTPHeader& rtpHeader);
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VCMPacket(const uint8_t* ptr,
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size_t size,
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uint16_t seqNum,
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uint32_t timestamp,
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bool markerBit);
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void Reset();
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uint8_t payloadType;
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uint32_t timestamp;
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// NTP time of the capture time in local timebase in milliseconds.
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int64_t ntp_time_ms_;
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uint16_t seqNum;
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const uint8_t* dataPtr;
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size_t sizeBytes;
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bool markerBit;
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FrameType frameType;
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VideoCodecType codec;
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bool isFirstPacket; // Is this first packet in a frame.
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VCMNaluCompleteness completeNALU; // Default is kNaluIncomplete.
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bool insertStartCode; // True if a start code should be inserted before this
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// packet.
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int width;
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int height;
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RTPVideoHeader codecSpecificHeader;
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protected:
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void CopyCodecSpecifics(const RTPVideoHeader& videoHeader);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
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