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}
155 lines
4.1 KiB
C++
155 lines
4.1 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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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/modules/video_coding/main/source/packet.h"
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#include <assert.h>
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namespace webrtc {
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VCMPacket::VCMPacket()
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: payloadType(0),
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timestamp(0),
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ntp_time_ms_(0),
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seqNum(0),
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dataPtr(NULL),
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sizeBytes(0),
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markerBit(false),
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frameType(kEmptyFrame),
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codec(kVideoCodecUnknown),
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isFirstPacket(false),
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completeNALU(kNaluUnset),
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insertStartCode(false),
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width(0),
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height(0),
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codecSpecificHeader() {}
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VCMPacket::VCMPacket(const uint8_t* ptr,
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const size_t size,
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const WebRtcRTPHeader& rtpHeader) :
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payloadType(rtpHeader.header.payloadType),
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timestamp(rtpHeader.header.timestamp),
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ntp_time_ms_(rtpHeader.ntp_time_ms),
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seqNum(rtpHeader.header.sequenceNumber),
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dataPtr(ptr),
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sizeBytes(size),
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markerBit(rtpHeader.header.markerBit),
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frameType(rtpHeader.frameType),
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codec(kVideoCodecUnknown),
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isFirstPacket(rtpHeader.type.Video.isFirstPacket),
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completeNALU(kNaluComplete),
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insertStartCode(false),
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width(rtpHeader.type.Video.width),
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height(rtpHeader.type.Video.height),
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codecSpecificHeader(rtpHeader.type.Video)
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{
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CopyCodecSpecifics(rtpHeader.type.Video);
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}
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VCMPacket::VCMPacket(const uint8_t* ptr,
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size_t size,
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uint16_t seq,
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uint32_t ts,
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bool mBit) :
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payloadType(0),
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timestamp(ts),
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ntp_time_ms_(0),
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seqNum(seq),
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dataPtr(ptr),
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sizeBytes(size),
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markerBit(mBit),
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frameType(kVideoFrameDelta),
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codec(kVideoCodecUnknown),
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isFirstPacket(false),
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completeNALU(kNaluComplete),
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insertStartCode(false),
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width(0),
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height(0),
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codecSpecificHeader()
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{}
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void VCMPacket::Reset() {
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payloadType = 0;
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timestamp = 0;
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ntp_time_ms_ = 0;
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seqNum = 0;
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dataPtr = NULL;
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sizeBytes = 0;
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markerBit = false;
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frameType = kEmptyFrame;
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codec = kVideoCodecUnknown;
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isFirstPacket = false;
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completeNALU = kNaluUnset;
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insertStartCode = false;
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width = 0;
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height = 0;
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memset(&codecSpecificHeader, 0, sizeof(RTPVideoHeader));
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}
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void VCMPacket::CopyCodecSpecifics(const RTPVideoHeader& videoHeader) {
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if (markerBit) {
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codecSpecificHeader.rotation = videoHeader.rotation;
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}
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switch (videoHeader.codec) {
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case kRtpVideoVp8:
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// Handle all packets within a frame as depending on the previous packet
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// TODO(holmer): This should be changed to make fragments independent
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// when the VP8 RTP receiver supports fragments.
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if (isFirstPacket && markerBit)
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completeNALU = kNaluComplete;
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else if (isFirstPacket)
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completeNALU = kNaluStart;
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else if (markerBit)
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completeNALU = kNaluEnd;
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else
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completeNALU = kNaluIncomplete;
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codec = kVideoCodecVP8;
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return;
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case kRtpVideoVp9:
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if (isFirstPacket && markerBit)
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completeNALU = kNaluComplete;
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else if (isFirstPacket)
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completeNALU = kNaluStart;
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else if (markerBit)
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completeNALU = kNaluEnd;
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else
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completeNALU = kNaluIncomplete;
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codec = kVideoCodecVP9;
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return;
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case kRtpVideoH264:
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isFirstPacket = videoHeader.isFirstPacket;
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if (isFirstPacket)
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insertStartCode = true;
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if (isFirstPacket && markerBit) {
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completeNALU = kNaluComplete;
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} else if (isFirstPacket) {
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completeNALU = kNaluStart;
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} else if (markerBit) {
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completeNALU = kNaluEnd;
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} else {
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completeNALU = kNaluIncomplete;
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}
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codec = kVideoCodecH264;
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return;
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case kRtpVideoGeneric:
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case kRtpVideoNone:
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codec = kVideoCodecUnknown;
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return;
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}
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}
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} // namespace webrtc
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